Re: [asterisk-users] Unable to build Hpec on Zaptel 1.4.10.1
Andres: Did you install the hpec in /usr/zaptel/kernel/hpec ??? Thanks Ruben Andres escribió: Tzafrir Cohen wrote: On Fri, May 09, 2008 at 06:36:36PM -0400, Andres wrote: Hi, I just tried to upgrade from Zaptel 1.4.6 to 1.4.10.1 but it looks like the Makefile has no rules for the HPEC. make[3]: *** No rule to make target `hpec/hpec_zaptel.h', needed by `/usr/src/zaptel/kernel/zaptel-base.o'. Stop. Can somebody confirm this? Yes, it looks wrong. Could you please try replacing the line that has hpec_zaptel.h with: $(obj)/zaptel-base.o: $(src)/hpec/hpec_zaptel.h $(src)/hpec/hpec_user.h (Note the added '$(src)/' ) But the Makefile has no line with hpec. Look at Zaptel 1.4.10.1 [EMAIL PROTECTED] zaptel]# grep hpec Makefile [EMAIL PROTECTED] zaptel]# And Look at Zaptel 1.4.6 [EMAIL PROTECTED] zaptel146]# grep hpec Makefile ifneq ($(wildcard $(PWD)/hpec/hpec_x86_32.o_shipped),) ifneq ($(wildcard $(PWD)/hpec/hpec_x86_64.o_shipped),) ZAPTEL_HPEC:=hpec/hpec_x86_32.o_shipped ZAPTEL_HPEC:=hpec/hpec_x86_64.o_shipped KFLAGS+=-DECHO_CAN_HPEC -I$(PWD)/hpec zaptel-base.o: hpec/hpec_zaptel.h hpec/hpec_user.h It looks like it was left out completely on the latest Zaptel. Should I open the bug report? Thanks, Andres. ___ -- 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] Unable to build Hpec on Zaptel 1.4.10.1
Why you dont download zaptel branches.? svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zap-branches Andres escribió: Ruben Zamora wrote: Andres: Did you install the hpec in /usr/zaptel/kernel/hpec ??? I sure did. Without it there is no error as no hpec support is attempted. Andres Thanks Ruben Andres escribió: Tzafrir Cohen wrote: On Fri, May 09, 2008 at 06:36:36PM -0400, Andres wrote: Hi, I just tried to upgrade from Zaptel 1.4.6 to 1.4.10.1 but it looks like the Makefile has no rules for the HPEC. make[3]: *** No rule to make target `hpec/hpec_zaptel.h', needed by `/usr/src/zaptel/kernel/zaptel-base.o'. Stop. Can somebody confirm this? Yes, it looks wrong. Could you please try replacing the line that has hpec_zaptel.h with: $(obj)/zaptel-base.o: $(src)/hpec/hpec_zaptel.h $(src)/hpec/hpec_user.h (Note the added '$(src)/' ) But the Makefile has no line with hpec. Look at Zaptel 1.4.10.1 [EMAIL PROTECTED] zaptel]# grep hpec Makefile [EMAIL PROTECTED] zaptel]# And Look at Zaptel 1.4.6 [EMAIL PROTECTED] zaptel146]# grep hpec Makefile ifneq ($(wildcard $(PWD)/hpec/hpec_x86_32.o_shipped),) ifneq ($(wildcard $(PWD)/hpec/hpec_x86_64.o_shipped),) ZAPTEL_HPEC:=hpec/hpec_x86_32.o_shipped ZAPTEL_HPEC:=hpec/hpec_x86_64.o_shipped KFLAGS+=-DECHO_CAN_HPEC -I$(PWD)/hpec zaptel-base.o: hpec/hpec_zaptel.h hpec/hpec_user.h It looks like it was left out completely on the latest Zaptel. Should I open the bug report? Thanks, Andres. ___ -- 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 ___ -- 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] I hear noise in the line
Hi I have a little probelm with my ip phone and asterisk, i dont know where can i look? When i place a call o receive a call, after talk or the other side finish talk, we both side hear ss (noise). i have installed the last zaptel branches and the last asterisk branches, 6 digium card TDM800p and HPec Thanks Ruben ___ -- 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] How i know the version of my vpmadt032 firmware
Hi How can i know the version of my vpmadt032 firmware? Thanks Ruben ___ -- 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] How i know the version of my vpmadt032 firmware
Shaun And what it the last version of that firmware? Thanks Ruben Shaun Ruffell escribió: Ruben, Ruben Zamora wrote: How can i know the version of my vpmadt032 firmware? Currently, it is only printed in the kernel log (view with the command dmesg) when the driver is loaded. Shaun ___ -- 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] choopy audio when both side talk at the same time
Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ,AND when the both side talks at the same time i have choppy audio. Any help i appreciate. Thanks Ruben ___ -- 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] MFC/R2 in chan_zap , Testers Wanted
Moises Thats means, that we arent going to use unicall? If that true i can test these weekend with a E1-Axtel. Thanks Ruben Moises Silva escribió: If you are an MFC/R2 user and want to help in the development of chan_zap support for this signalling, please take a look at the bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact me. Currently just México support is built-in, if you want your country variant supported, drop me a line. Moisés Silva ___ -- 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] Problems with Quality Voice in a Asterisk-E1-Unicall
I Have with Asterisk- Unicall - E1 (MFC/R2). Days before a install a Digium Card TE122P with hardware echo cancelation, these because a had a echo in some in and out calls. I replaced the card. I no more echo but in my conversation the voice start to doing things. Like after a minutes i start hearing the voice cut. or the cant hearme.. I remove in the zaptel.conf the echotraining.I dont know if i really need to do these changes in the unicall.conf.??? In my Asterisk am using GXP2020 Grandstream what is better ulaw,alaw,g729??? I apreciate any help. Thanks Ruben ___ -- 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] tdm410p w/ echo - no full duplex
Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] tdm410p w/ echo - no full duplex
You can read these information in the zapata.conf. Most of the time when you use hardware cancelation echo these paramater make worse echo. Its better when you use HPEC that is a software no hardware for that parameter. Michael J. Liberatore escribió: Ok I will remove it, may I ask what that will do or how that will help? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Zamora Sent: Friday, April 11, 2008 7:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael Check your /etc/asterisk/zapata.conf and if you have echocancelwhenbridge=yes, remove Ruben Michael J. Liberatore escribió: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 ___ -- 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] Half-duplex call on TDM2400p with VPMADT032
Today I Install Zaptel 1.4.10 and compiled.No good result. Then Digium Support send me the last firmware of VPMADT032, and installed, at the first sight there was no good news. But then i move in the driver wcte12xp in the file base.c and i have better results. Matthew Fredrickson escribió: Faraz R. Khan wrote: The newer zaptel (1.4.10) says it includes firmware 1.16 from the CHANGELOG: firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay. We had to back that version of the firmware out due to release related problems. As for all problems related to the VPMADT032, if you have any issues, please contact technical support. They will be able to help you with whatever issue you may have. Matthew Fredrickson The URL provided does not contain firmware for the VPMADT032 I* have logged a query with digum. Is there a URL to get this firmware from? On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote: Lex Thanks, I all ready download the last svn branches from zaptel And i am going to test these afternoon. My phone number es 81-83481611. Thanks Ruben Lex Lethol escribió: Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- 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 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 ___ -- 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] Half-duplex call on TDM2400p with VPMADT032
Lex Thanks, I all ready download the last svn branches from zaptel And i am going to test these afternoon. My phone number es 81-83481611. Thanks Ruben Lex Lethol escribió: Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- 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 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 ___ -- 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] Half-duplex call on TDM2400p with VPMADT032
Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. By the moment i have a big problem. Thanks Ruben Lex Lethol escribió: Ruben, Contact support at digium they have a release on a firmware that fixes this and other issues with the VPMADT032. Apparently it comes on newer zaptel drivers. Good luck with your install. Lex On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ruben Zamora wrote: Hi, I have a same problem, last week i was working with TE120 with a little echo in some call, I replace the card with a TE122B ( Included Echo Cancelation VPMADT032) and there was no more echo in my call. But know i have de same probelm with my incoming audio stream gets clipped / dropped when you speak. Please contact Digium technical support about this. This is definitely something that we need to work with the vendor of the echo canceller IP about. Matthew Fredrickson Thanks Ruben Lex Lethol escribió: Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- 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 ___ -- 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] Half-duplex call on TDM2400p with VPMADT032
Hi, I have a same problem, last week i was working with TE120 with a little echo in some call, I replace the card with a TE122B ( Included Echo Cancelation VPMADT032) and there was no more echo in my call. But know i have de same probelm with my incoming audio stream gets clipped / dropped when you speak. Thanks Ruben Lex Lethol escribió: Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- 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] Problems with Unicall and TE122B
Hi By the past 2 months i have install a server with Asterisk with a E1 in Axtel(Mexico). The call presented a Echo in randoms call, inside and outside.I decide migrate for the last version Asterisk 18.1,Zaptel 1.49 and Unicall and instalaled a new Digim Card TE122 B (Echo cancelation). The echo goes away. But i stat gaving probelms with my incoming audio stream gets clipped / dropped when they speak. I dont know if i need to move a parameter in unicall.conf, or in another file. Thanks Ruben ___ -- 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] Problems with DELL 1600
Hi I just want to know if anyone have problems with server DELL 1600, Like: Hangup Call. Thanks Ruben ___ -- 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] G.729 Copy Protection
Jacobson escribió: Hi, Did you contact digium ? I did contact them some years ago with a similar problem (but different) and they were helpful. Good luck *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Guilherme Loch Waltrick Góes *Sent:* Monday 24 March 2008 17:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] G.729 Copy Protection But the problem is, I haven't changed any hardware. I just reinstalled my server. Anyone from Digium can help me with this ? Best Regards, On Mon, Mar 24, 2008 at 1:28 PM, Eric Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Looks to me like you changed the ethernet controller. The G729 copy protection is based on the MAC of the interfaces in the system. Guilherme Loch Waltrick Góes wrote: I'm trying to use the Digium suplied G.729 Codec, I have ran the register utility, and got my licenses written to /var/lib/asterisk/licenses, but when a start Asterisk I got the following errors: [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: Please see the full license text supplied by the accompanying [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: register utility, or ask for a copy from Digium. [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: This product includes software developed by the OpenSSL Project [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Mar 24 12:33:08] NOTICE[4068] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [Mar 24 12:33:08] VERBOSE[4068] logger.c: == G.729 Host-ID: 'MY MAC ADDRESS' [Mar 24 12:33:08] VERBOSE[4068] logger.c: == Found license 'MY LICENSE' providing 6 channels [Mar 24 12:33:08] WARNING[4068] codec_g729a.c: Failed to initialize G.729copy protection! [Mar 24 12:33:08] VERBOSE[4068] logger.c: codec_g729a.so = (Annex A/B (floating point) G.729 Coder/Decoder (optimized for i686)) [Mar 24 12:33:13] VERBOSE[4068] logger.c: == Registered file format g729, extension(s) g729 [Mar 24 12:33:13] VERBOSE[4068] logger.c: format_g729.so = (Raw G729 data) in /var/log/asterisk/full. What can I do ? ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.8/1340 - Release Date: 23/03/2008 18:50 ___ -- 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 Did you backup your /var/lib/asterisk/licenses ? If you reinstall your machine i didnt backup the licenses you need to contact Digium support. They can give you access to reinstall your G729 licenses. ___ -- 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] Little help with Conference
These is my scenario. Asterisk 1.4.16 Zaptel1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make reverse for these??? Thanks Ruben ___ -- 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] Problem with IRQ Share
Hi I have a Server with Centos 5, TDM400p, HP Server ML110. My problem is that I see IRQ Share with my TDM400P. How can I fix that??? ___ -- 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] Echo in the outside call (E1)
I have a little problem with mi outside calls. I Have Echo. It`s randomly. I install codec G729 and I have Grandstream GXP2020. Asterisk 1.4.9,Unicall 0.0.5pre1,Zaptel 1.4.5 I have a E1 and 30 channel. Can you give a tip, where can I check If I miss something. Thanks ___ -- 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] Problem with TDM400P
I have install a Asterisk 1.4.9 with Centos, a TDM400P (4 Analog Lines) my problem is one o two day a week one of the lines have a lot of noise, I i cant place a call outside.I need to reboot the server to get the lines again. Do you know is it's another way to check the lines o reset the port in the Digium card??? Thanks Ruben Zamora ___ -- 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] Problem with a channel
I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5 I having these problem : Zap/2-1 is busy Hangup ZAP/2-1 Everyone is busy/congested at this time (1:1/010) Autofallthrough channel SIP/202-b7b08ab0 Status is busy. And then HANGUP. ___ -- 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] Problem with a channel
The problem is that i have random hangup in calls in the PSTN. After that I check in asterisk -rvv Sip show channels And I see the extension The only way that I can place another call in the extension was to restart the Asterisk. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Moises Silva Enviado el: Miércoles, 16 de Enero de 2008 09:31 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Problem with a channel And the problem is? ... I think you should read this: http://catb.org/~esr/faqs/smart-questions.html Regards, Moisés Silva On Jan 16, 2008 6:42 PM, Ruben Zamora [EMAIL PROTECTED] wrote: I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5 I having these problem : Zap/2-1 is busy Hangup ZAP/2-1 Everyone is busy/congested at this time (1:1/010) Autofallthrough channel SIP/202-b7b08ab0 Status is busy. And then HANGUP. ___ -- 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 -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ -- 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
