Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
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. 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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
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 -- 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
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
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
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
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
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
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
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
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
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] Half-duplex call on TDM2400p with VPMADT032
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
