Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061030/1235e58c/attachment.htm
Alberto Pastore
2006-Oct-30 23:56 UTC
[asterisk-users] Cisco 7960 Skinny calling SIP phone
Well, I've never actually been able to make chan_skinny work with 79xx phones. I found the chan_sccp to work quite well: http://chan-sccp.berlios.de/ plus this patch for a problem on MeetMe (I don't remeber where I found it, but it works!): diff -uNr chan_sccp-20060408.org/sccp_pbx.c chan_sccp-20060408/sccp_pbx.c --- chan_sccp-20060408.org/sccp_pbx.c 2006-04-08 14:20:17.000000000 +0200 +++ chan_sccp-20060408/sccp_pbx.c 2006-05-17 17:14:15.000000000 +0200 @@ -290,6 +290,12 @@ static int sccp_pbx_answer(struct ast_channel *ast) { sccp_channel_t * c = CS_AST_CHANNEL_PVT(ast); + // if channel type is undefined, set to SCCP + if (!ast->type) { + sccp_log(1)(VERBOSE_PREFIX_3 "SCCP: Channel type undefined, sett ing to type 'SCCP'\n"); + ast->type = "SCCP"; + } + if (!c || !c->device || !c->line) { ast_log(LOG_ERROR, "SCCP: Answered %s but no SCCP channel\n", as t->name); return -1; I recommend using SIP firmware anyway... the conversion process is a bit annoying but as far as now 7940/7960 are really stable IP phones. I am currently using chan_sccp only for 7902 phones (I've just got 2 of them....) which do not support SIP firmware. Will Roy ha scritto:> Before I got down the path of converting a Cisco 7960 I have over to > SIP I wanted to try and set it up using Skinny. > > The phone registers ok with Asterisk. When I call a SIP softphone > extension on my network the call is made and I can answering it. > However no voice is heard over the call. > > When I debug Skinny on the console after the call has connected I see > the following messag: > > Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] > > What additional information would be required to troubleshoot this? or > should I stop wasting time and just convert the phone to SIP? :) > > regards > Wil > > ------------------------------------------------------------------------ > > _______________________________________________ > --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 >-- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it
Anthony LaMantia
2006-Oct-31 09:57 UTC
[asterisk-users] Cisco 7960 Skinny calling SIP phone
Which asterisk release are you running chan_skinny under? ----- Original Message ----- From: Will Roy <willroyvi@gmail.com> To: asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil
I am running 1.4.0-beta2 Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST) From: Anthony LaMantia <alamantia@digium.com> Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <18975956.15221162313826243.JavaMail.root@jupiler.digium.com> Content-Type: text/plain; charset=utf-8 Which asterisk release are you running chan_skinny under? ----- Original Message ----- From: Will Roy <willroyvi@gmail.com> To: asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061031/83036d19/attachment.htm