RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Well I just took a look at the TAC case and things dont look good, seems the TAC are now blaming Asterisk for the problem but I will go through there debugs and push back, will let you know. -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 08 March 2004 22:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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
RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Adam, Does Polycom license the SIP stuff from Cisco? If not, then Asterisk may be the culprit, because all of my Polycom IP500s exhibit the same behavior. I'm running asterisk 0.7.1, Zaptel CVS and libpri CVS both from a few days ago, but I don't recall having this problem a few months ago when I was running older versions. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Low, Adam Sent: Wednesday, March 10, 2004 1:03 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Well I just took a look at the TAC case and things dont look good, seems the TAC are now blaming Asterisk for the problem but I will go through there debugs and push back, will let you know. -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 08 March 2004 22:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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 ___ 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
Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ 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
RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ 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 * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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
