Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
Andrew Gillham wrote: Sounds good. I have not been that bothered with it when I make a normal voice call. It is mostly annoying when hitting the messages button on the phone. My delay helped that situation. Perhaps on calls where asterisk is proxying the rtp stream we could have an option to tell asterisk to open the connection to the 7960 before the connection is setup on the other side of the call. So the 7960 gets a head start. It would force the codec but that is fine by me, my G.729 is preferred and I don't mind asterisk transcoding since I have a low number of calls. -Andrew I think Barton found the root problem, Native bridging fails or takes to long to setup causeing the delay. I am going to see if a bug has already been opened on this and if not do so. ___ 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 starts after ring.
I hacked the Wait command to wait in increments of 100ms. The 7960 needs about 300ms delay after answer to play the sound properly. ATA186's work fine without any delay for me. A finer grained 'Wait' would be helpful in developing workarounds for this sort of problem. Iain --On Wednesday, March 10, 2004 6:04 pm -0800 Andrew Gillham [EMAIL PROTECTED] wrote: Steve Creel wrote: 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... exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew ___ 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 starts after ring.
On Thu, 11 Mar 2004, Iain Stevenson wrote: I hacked the Wait command to wait in increments of 100ms. The 7960 needs about 300ms delay after answer to play the sound properly. ATA186's work fine without any delay for me. A finer grained 'Wait' would be helpful in developing workarounds for this sort of problem. As of 3/4/2004 in cvs head and stable the Wait application has accepted time with fractions of a second. So 0.1 would be 100ms, 0.3 would be 300ms, etc. James ___ 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 starts after ring.
--On Thursday, March 11, 2004 3:17 am -0500 James Golovich [EMAIL PROTECTED] wrote: As of 3/4/2004 in cvs head and stable the Wait application has accepted time with fractions of a second. So 0.1 would be 100ms, 0.3 would be 300ms, etc. James Thanks, that makes a workaround for the 7960 problem this: exten = 40,1,Answer exten = 40,2,Wait,0.3 exten = 40,3,VoicemailMain2 Iain ___ 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 starts after ring.
exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. ___ 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 starts after ring.
[EMAIL PROTECTED] wrote: exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. I've seen the same thing, and it appears to be from attempting a native bridge. You can try the attached patch to disable native bridging. It cut out the annoying silence completely for me. This may be a bad thing (unnecessary CPU utilization due to same-codec translation), but I have not experienced any problems. Barton channel.c.diff Description: Binary data
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
James Sizemore wrote: exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. According to the bug release notes this is caused by the DSP setup on the 7960. I would guess that it must need to setup the correct codec once it is selected and that takes some time (400ms apparently). Perhaps they could create a 'leave the dsp setup for codec X and never change codecs' config option. :-) This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. Sounds good. I have not been that bothered with it when I make a normal voice call. It is mostly annoying when hitting the messages button on the phone. My delay helped that situation. Perhaps on calls where asterisk is proxying the rtp stream we could have an option to tell asterisk to open the connection to the 7960 before the connection is setup on the other side of the call. So the 7960 gets a head start. It would force the codec but that is fine by me, my G.729 is preferred and I don't mind asterisk transcoding since I have a low number of calls. -Andrew ___ 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 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:[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 starts after ring.
What versions of Zaptel, Asterisk, and libpri? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Fraizer Sent: Wednesday, March 10, 2004 2:08 PM To: [EMAIL PROTECTED] 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:[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 ___ 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 starts after ring.
Bisker, Scott (7805) wrote: What versions of Zaptel, Asterisk, and libpri? I downloaded them all at the same time from CVS. I really couldn't tell you though off the top of my head. John ___ 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 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 [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 starts after ring.
I have experienced this behavior on the 7960 as well. - Chris - Original Message - From: Steve Creel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 3:18 PM 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 [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 starts after ring.
Steve Creel wrote:
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
exten = 8500,1,Answer
exten = 8500,2,Ringing
exten = 8500,3,Wait,1
exten = 8500,4,VoicemailMain(${CALLERIDNUM})
exten = 8500,5,Hangup
Works fine.
If you're talking about the comedian mail cutting off at the beginning
without having the answer,ringing,wait in there, I think you're being a bit
picky, perhaps even anal about the timing there.
It boils down to the fact that voicemailmain doesn't wait to see the session
completely up before playing the comedian mail prompt. This is indeed an
Asterisk issue - one that is very easy to remedy by setting your voicemail
extension as I have mine.
For what it's worth, my Grandstream phone has the same exact behavior
without the answer,ringing,wait.
What exactly happens in that first 5ms of call setup that is so crucial?
I've seen PSTN switches with longer delays in setup/teardown.
John
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Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
Steve Creel wrote: 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... exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew ___ 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 starts after ring.
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). For my first two or three months of using Asterisk I had this problem (with a Cisco 1750 and Cisco FXO and FXS cards). I thought the name of Asterisk's voicemail system was Median Mail, which I believe is a copyright/tradmatk of Northern Telecom. I stoped using the Cisco voice cards when the TDM400P was released. I've not experienced this problem with Cisco ATA-186 SIP firmware 1.6.1 or Cisco 7905G firmware 1.0.1. I don't know if the firmware updates fixed the problem or something else fixed the problem. ___ 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] 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? ___ 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 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 ___ 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 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, March 03, 2004 8:11 AM To: [EMAIL PROTECTED] 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 [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 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
