James Sizemore
2004-Mar-03 05:55 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
When calling out on a Cisco 7960 there is a short delay before the call gets setup and the other side can hear your voice. Anyone know how to compensate for this effect?
Lele Forzani
2004-Mar-03 06:10 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
On Wednesday 03 March 2004 13:55, James Sizemore wrote:> When calling out on a Cisco 7960 there is a short delay before the call > gets setup and the other side can hear your voice. > Anyone know how to compensate for this effect?Open Caveats Release 6.2 This section documents possible unexpected behavior by Cisco IP Phone 7940/7960 Release 6.2. This section lists only severity 1 and 2 caveats and select severity 3 caveats. CSCed40056: SIPPhone: DND config causes weird NTP behavior CSCed48311: Media takes 0.4 sec to be set up bye lele
Rich Adamson
2004-Mar-03 06:10 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
> When calling out on a Cisco 7960 there is a short delay before the call > gets setup and the other side can hear your voice. > Anyone know how to compensate for this effect?Sounds like the 7960 has not been configured with a dialplan that supports your * dialplan. Look for the dialplan.xml file on your tftp server and check its contents. Should look something like the following: <DIALTEMPLATE> <TEMPLATE MATCH="0" Timeout="1" User="Phone"/> <!-- Local operator--> <TEMPLATE MATCH="911" Timeout="0" User="Phone"/> <!-- Local numbers--> <TEMPLATE MATCH="3..." Timeout="0" User="Phone"/> <!-- Corporate Dial plan--> <TEMPLATE MATCH="4,4......" Timeout="0" User="Phone"/> <!-- Local numbers--> <TEMPLATE MATCH="5,4......" Timeout="0" User="Phone"/> <!-- Local numbers--> </DIALTEMPLATE> The first entry, above, says if the user dialed "0", then wait for one second to ensure they didn't dial something like "0-555-1212". If no other digits dialed, the 7960 is supposed to send "0" to asterisk after that 1-second timeout. The third entry says my local * extensions are four-digit numbers starting with a "3". If the user dial 3111, the 7960 should immediately send that to * (no timeout). Rich
Bisker, Scott (7805)
2004-Mar-03 07:04 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
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. SIP phone dials 555-1234 (outside line via PRI) 555-1234 rings 555-1234 answers and says "Hello" SIP phone hears "o" or nothing at all. If 555-1234 is slow to say something, then everything is heard fine. Caveats. echotraining and echocancel are enabled on the PRI, however, similiar Zap calls are not affected. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Rich Adamson Sent: Wednesday, March 03, 2004 8:11 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.> When calling out on a Cisco 7960 there is a short delay before the call > gets setup and the other side can hear your voice. > Anyone know how to compensate for this effect?Sounds like the 7960 has not been configured with a dialplan that supports your * dialplan. Look for the dialplan.xml file on your tftp server and check its contents. Should look something like the following: <DIALTEMPLATE> <TEMPLATE MATCH="0" Timeout="1" User="Phone"/> <!-- Local operator--> <TEMPLATE MATCH="911" Timeout="0" User="Phone"/> <!-- Local numbers--> <TEMPLATE MATCH="3..." Timeout="0" User="Phone"/> <!-- Corporate Dial plan--> <TEMPLATE MATCH="4,4......" Timeout="0" User="Phone"/> <!-- Local numbers--> <TEMPLATE MATCH="5,4......" Timeout="0" User="Phone"/> <!-- Local numbers--> </DIALTEMPLATE> The first entry, above, says if the user dialed "0", then wait for one second to ensure they didn't dial something like "0-555-1212". If no other digits dialed, the 7960 is supposed to send "0" to asterisk after that 1-second timeout. The third entry says my local * extensions are four-digit numbers starting with a "3". If the user dial 3111, the 7960 should immediately send that to * (no timeout). Rich _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Bisker, Scott (7805)
2004-Mar-10 12:31 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
What versions of Zaptel, Asterisk, and libpri? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of John Fraizer Sent: Wednesday, March 10, 2004 2:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. For what it's worth, I don't have any delay between answer and audio with my asterisk server and 7960G either originating or answering. It doesn't matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's pretty much instant (not detectable by humans at least). So, there may be some truth to the fact that the delay is caused by the Asterisk install in your case. There are so many variables that it is very hard to tell but, since I don't see the delay, I am leaning towards it being an Asterisk implementation issue. Here's what I'm running: Compaq DL380 1Gha with 1GB of memory Redhat Linux 8.0 (soon to be Gentoo - amazing difference in performance) Asterisk version: CVS-02/15/04-14:03:51 7960 Firmware Version: Application Load ID = P0S3-06-1-00 Boot Load ID = PC030301 DSP Load ID = PS03AT38 I'm using the ULAW codec. John Low, Adam wrote:> 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:james@deny.org] > Sent: 08 March 2004 22:09 > To: asterisk-users@lists.digium.com > 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:digium@aus-biz.com] >>Sent: 03 March 2004 15:12 >>To: asterisk-users@lists.digium.com >>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 > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Bisker, Scott (7805)
2004-Mar-10 13:27 UTC
[Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
Same behavior here. IP500 and 7960G phones cutoff first part of VoiceMailMain. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Steve Creel Sent: Wednesday, March 10, 2004 3:18 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. On Wed, 10 Mar 2004, John Fraizer wrote:> >For what it's worth, I don't have any delay between answer and audio with my > asterisk server and 7960G either originating or answering. It doesn't >matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's >pretty much instant (not detectable by humans at least). So, there may be >some truth to the fact that the delay is caused by the Asterisk install in >your case. There are so many variables that it is very hard to tell but, >since I don't see the delay, I am leaning towards it being an Asterisk >implementation issue.Can you test this with an extension that goes into VoiceMailMain(). My 7960 and 7960G phones both get the first couple letters of "Commedian Mail" cut off (usually "...median Mail"). Just trying to quantify the delay we're talking about... Steve _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users