Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
Vic, The problem you're having has been discussed multiple times on this list, and can be easily seen using ethereal to inspect the timestamps contained within the rtp packets sent "to" the 7960 phone. There are several issues involved, including: 1. the cisco phones drop any rtp packet that is not exactly 160 milliseconds between successive packets (thus causing choppy audio). That drop function seems to be the result of cisco changing DSPs in their v6.x code. I've not heard of anyone running v5.x sip code with the problem. 2. iax2 had a bug in it that Mark fixed last month. The bug resulted in iax2/gsm timestamps that were erratic when they should have been exactly 20 milliseconds between successive packets. 3. Code was added to rtp.c about a month or so ago that "ties" the iax2/gsm timestamps directly to the sip/rtp timestamps. When that code was added, it made the iax2 erratic timestamps "and" Cisco's dropping of packets extemely obvious to iax2 users. Other non-iax2 users are not impacted by this. Cisco phones seem to be the only ones impacted by this. There are three short term fixes available to you: a. upgrade (or insist your service provider) upgrade their iax2 code. (I don't believe the Stable branch has the fix in it as yet.) NuFone and some others have done that a few weeks ago. b. remove the two or three lines that were added in rtp.c (although Mark is discouraging this approach for other reasons), or, go back to source code cvs from about early March. c. Change the 7960's from v6.x code to v5.x code (and open a trouble ticket with Cisco). I've not heard anyone suggest that dropping rtp packets with uneven timestamps is necessary, a standard, or anything else. Therefore believe it's an anomaly that crept in with the DSP change in the sip v6.x code from Cisco. Rich > On Fri, 7 May 2004, Brian Cuthie wrote: > > > It seems that each time I get a new checkout of * from CVS my Cisco 7960 > > works worse than before. I know this stuff's in flux, so I mention this > > in case it's news. Anyone else having trouble? What I'm seeing (er, > > hearing) is really choppy audio. The previous version I had installed > > had fairly frequent audio dropouts (not present when I make the same > > calls through the same * box using a TDM400P interface). > > I had jittery audio with dropouts on a 7960 with SCCP, and started testing > SIP hoping it would be better (based on the reports of the SIP-to-IAX2 > timestamping issue). Here's my experience: > > * As Brian mentions, when the other end of the call is from a non-VoIP > path (e.g. Zaptel interface) the audio is fine. > > * Calls over IAX start out okay, but within a few seconds the audio starts > jittering. It gets progressively worse until about a minute into the call > (often less), by which time audio is unintelligible. Calling the same > number over the same IAX connection from an analogue phone attached to a > SIP-image ATA-186 which in turn is plugged into the "PC" port of the same > 7960 gives perfect audio. > > * Calls over SIP are stable; I had an intermittent problem where audio > into the 7960 would stop completely for up to three seconds, but that > seems to be gone after doing a CVS update. Side note: when I had this > audio dropout problem, making the same call without * in the audio path > (by using canreinvite=yes and removing t and T from Dial) resulted in > perfect audio. > > I'll try someone's suggestion to disable the jitterbuffer to fix the IAX2 > problem, but I thought that the jitterbuffer was supposed to help this > kind of problem... Besides, the same call over an ATA or using X-lite is > perfect. > > Before anyone jumps in, yes, as soon as I can get there I will hit the bug > tracker. > > Cheers, > Vic Cross > > > PS: I know that folks generally dislike 'me too' messages, but this time > Too Bad -- I'm trying to provide more info to help anyone that might be > working on problems. > > > I hope that Iain was exaggerating when he described his bug-reporting > experience. Many * users are unable to commit the time to poring over > hundreds of lines of uncommented C code and ethereal traces with thousands > of packets captured. So, as our way of trying to help, we provide e-mails > like this either in response to or as an attempt to gather more > information about the problem. To try and get people talking about a > problem. > > How is does it help to jump on someone who is trying to get resolution to > a problem -- by driving them toward OpenPBX or VOCAL? A few former > colleagues of mine may soon be about to learn (unfortunately) that you can > only piss off a customer so many times. > > To the Asterisk developers, bug marshals, and coders: I am jealous of you! > You've created a wonderful thing. I'd love to be able to spend the amount > of time I'd like to on Asterisk. I'd love to be able to do
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
On Fri, 7 May 2004, Brian Cuthie wrote: > It seems that each time I get a new checkout of * from CVS my Cisco 7960 > works worse than before. I know this stuff's in flux, so I mention this > in case it's news. Anyone else having trouble? What I'm seeing (er, > hearing) is really choppy audio. The previous version I had installed > had fairly frequent audio dropouts (not present when I make the same > calls through the same * box using a TDM400P interface). I had jittery audio with dropouts on a 7960 with SCCP, and started testing SIP hoping it would be better (based on the reports of the SIP-to-IAX2 timestamping issue). Here's my experience: * As Brian mentions, when the other end of the call is from a non-VoIP path (e.g. Zaptel interface) the audio is fine. * Calls over IAX start out okay, but within a few seconds the audio starts jittering. It gets progressively worse until about a minute into the call (often less), by which time audio is unintelligible. Calling the same number over the same IAX connection from an analogue phone attached to a SIP-image ATA-186 which in turn is plugged into the "PC" port of the same 7960 gives perfect audio. * Calls over SIP are stable; I had an intermittent problem where audio into the 7960 would stop completely for up to three seconds, but that seems to be gone after doing a CVS update. Side note: when I had this audio dropout problem, making the same call without * in the audio path (by using canreinvite=yes and removing t and T from Dial) resulted in perfect audio. I'll try someone's suggestion to disable the jitterbuffer to fix the IAX2 problem, but I thought that the jitterbuffer was supposed to help this kind of problem... Besides, the same call over an ATA or using X-lite is perfect. Before anyone jumps in, yes, as soon as I can get there I will hit the bug tracker. Cheers, Vic Cross PS: I know that folks generally dislike 'me too' messages, but this time Too Bad -- I'm trying to provide more info to help anyone that might be working on problems. I hope that Iain was exaggerating when he described his bug-reporting experience. Many * users are unable to commit the time to poring over hundreds of lines of uncommented C code and ethereal traces with thousands of packets captured. So, as our way of trying to help, we provide e-mails like this either in response to or as an attempt to gather more information about the problem. To try and get people talking about a problem. How is does it help to jump on someone who is trying to get resolution to a problem -- by driving them toward OpenPBX or VOCAL? A few former colleagues of mine may soon be about to learn (unfortunately) that you can only piss off a customer so many times. To the Asterisk developers, bug marshals, and coders: I am jealous of you! You've created a wonderful thing. I'd love to be able to spend the amount of time I'd like to on Asterisk. I'd love to be able to do more to fix bugs and develop features. But I can't. Don't think less of me because of that. VC ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
This isn't really the issue. Up until a week ago or so everything worked fine with a hallf duplex hub. Now it doesn't - so I suspect some code change made in * is responsible. I think * must maintain backwards compatibility with existing hardware or many people will get fed up with constant degradation of sound quality. Iain --On Friday, May 07, 2004 14:15:47 -0600 James Sizemore <[EMAIL PROTECTED]> wrote: I checked-out CVS Head today to get realm support, I have over hundred Cisco phone on my servers and I have not noticed any Qos problems. You may want to check the duplex of your switches and Asterisk boxes. If you don't have full duplex, that is more then likely your problem. Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
I checked-out CVS Head today to get realm support, I have over hundred Cisco phone on my servers and I have not noticed any Qos problems. You may want to check the duplex of your switches and Asterisk boxes. If you don't have full duplex, that is more then likely your problem. Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
This caught me as well. Be aware that if you did any manual mods to rtp.c or related files, you need to delete it and rerun 'make update'. This will bring down the proper file. You should then be all set. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 644 6999 x2002 > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Iain Stevenson > Sent: May 7, 2004 11:13 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 > problems persist (for me, anyway) > > > > I've had this too, reported it as a bug last week and got my > butt kicked > for not being responsive enough in providing support to sort > it out. You > could file another bug report but be sure to have a thick > book ready to > stuff down your trousers. > > Iain > > > --On Friday, May 7, 2004 10:43 am -0400 Brian Cuthie > <[EMAIL PROTECTED]> > wrote: > > > > > It seems that each time I get a new checkout of * from CVS > my Cisco 7960 > > works worse than before. I know this stuff's in flux, so I > mention this > > in case it's news. Anyone else having trouble? What I'm > seeing (er, > > hearing) is really choppy audio. The previous version I had > installed had > > fairly frequent audio dropouts (not present when I make the > same calls > > through the same * box using a TDM400P interface). > > > > Cheers, > > > > Brian > > ___ > > 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 > > -- > Scanned for viruses and dangerous content at > http://www.oneunified.net and is believed to be clean. > > -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
Do you have a jitterbuffer enabled on your inter-asterisk IAX trunks? If so, try disabling it cleared everything up for me. With jitter buffer enabled using the default settings my audio across the IAX trunk was terrible. BTW, my 7960's are running 5.3 firmware so I probably don't see the timestamp sensitive 6.x packet drops that have been discussed here. Bill Brian Cuthie <[EMAIL PROTECTED]> Brian Cuthie <[EMAIL PROTECTED]> Sent by: [EMAIL PROTECTED] 05/07/2004 08:57 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) Ah, this reminds me that I forgot to mention that our network looks like this: Cisco <--- SIP > Asterisk < IAX > Aterisk < IAX > Asterisk < PRI > PSTN -brian Tom wrote: > At 09:43 AM 5/7/2004, you wrote: > >> It seems that each time I get a new checkout of * from CVS my Cisco >> 7960 works worse than before. I know this stuff's in flux, so I >> mention this in case it's news. Anyone else having trouble? What >> I'm seeing (er, hearing) is really choppy audio. The previous version >> I had installed had fairly frequent audio dropouts (not present when >> I make the same calls through the same * box using a TDM400P interface). > > > No dropout problems or choppy audio running Asterisk > CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz > P4 Supermicro server. Analog phones through our TDM400P do sound much > better but the audio problems on our Cisco SIP phones are echo > problems. People are working on solutions. > > Tom > >> Cheers, >> >> Brian >> ___ >> 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
Upgrade each asterisk (iax2) and the problem will go away. As bkw mentioned, the problem "sources" from the location with the older iax2 code (which probably includes the Stable cvs I believe). NuFone had the problem in mid/late April as well, but they apparently updated their code when the issue was discovered/corrected. Other iax2 providers are likely to source the problem as well. Will take awhile for everyone to get the code into their production machines. > Ah, this reminds me that I forgot to mention that our network looks like > this: > > Cisco <--- SIP > Asterisk < IAX > Aterisk < IAX > > Asterisk < PRI > PSTN > > -brian > > > Tom wrote: > > > At 09:43 AM 5/7/2004, you wrote: > > > >> It seems that each time I get a new checkout of * from CVS my Cisco > >> 7960 works worse than before. I know this stuff's in flux, so I > >> mention this in case it's news. Anyone else having trouble? What > >> I'm seeing (er, hearing) is really choppy audio. The previous version > >> I had installed had fairly frequent audio dropouts (not present when > >> I make the same calls through the same * box using a TDM400P interface). > > > > > > No dropout problems or choppy audio running Asterisk > > CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz > > P4 Supermicro server. Analog phones through our TDM400P do sound much > > better but the audio problems on our Cisco SIP phones are echo > > problems. People are working on solutions. > > > > Tom > > > >> Cheers, > >> > >> Brian > >> ___ > >> 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 ---End of Original Message- ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
Just an FYI if you can run tethereal -n udp port 4569 And watch the timestamps(should be even 20ms increments per call leg). Both ends will need to be updated also. If not you will get some very strange timestamp issues and jitter and timestamps "might" not be right. If you have one end on cvs-stable and one on cvs-head you might see this problem also. I don't see any issues with IAX2 from my 7960 out nufone. bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Rich Adamson > Sent: Friday, May 07, 2004 11:36 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist > (for me, anyway) > > > It seems that each time I get a new checkout of * from CVS my Cisco 7960 > > works worse than before. I know this stuff's in flux, so I mention this > > in case it's news. Anyone else having trouble? What I'm seeing (er, > > hearing) is really choppy audio. The previous version I had installed > > had fairly frequent audio dropouts (not present when I make the same > > calls through the same * box using a TDM400P interface). > > Brian, > > Are you having the choppy audio only on iax2 links or on other calls as > well? > > There was an issue with erratic iax2 timestamps which caused the Cisco > phones to effectively drop any sip packet that had uneven timestamps > causing extremely choppy audio. The choppy audio (as I seen it) was > only in one direction (from the iax2 source with the erratic timestamps > towards to 7960 phone). > > If your issue is not associated with iax2, then be aware the Cisco v6.x > code changed DSP firmware internally, and any sip/rtp packets arriving > with > uneven timestamps (within the rtp pkts) will be dropped and cause the > choppy audio. You should be able to see the timestamps with ethereal. > The timestamp difference between successive pkts should be exactly > 160 milliseconds; if its anything else, the phone will drop the pkt. > (Not sure if that is a real Cisco bug or a planned change, but it > certainly has a hugh negative impact on voice quality.) > > I'm running CVS-HEAD-05/02/04 with no problems today. > > 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
Ah, this reminds me that I forgot to mention that our network looks like this: Cisco <--- SIP > Asterisk < IAX > Aterisk < IAX > Asterisk < PRI > PSTN -brian Tom wrote: At 09:43 AM 5/7/2004, you wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro server. Analog phones through our TDM400P do sound much better but the audio problems on our Cisco SIP phones are echo problems. People are working on solutions. Tom Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
I've got Asterisk STABLE-CVS-4/19/04 with 12 Cisco 7960 phones 6.0 Firmware using ulaw, 6 Polycom IP500 ulaw phones, and 192 Zap channels. I have Gig-E Copper to my server and 100Mbit-Full to all my phones. I haven't had any choppy audio at all. My switch is a Cisco 4500. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Sent: Friday, May 07, 2004 11:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) At 09:43 AM 5/7/2004, you wrote: >It seems that each time I get a new checkout of * from CVS my Cisco 7960 >works worse than before. I know this stuff's in flux, so I mention this in >case it's news. Anyone else having trouble? What I'm seeing (er, >hearing) is really choppy audio. The previous version I had installed had >fairly frequent audio dropouts (not present when I make the same calls >through the same * box using a TDM400P interface). No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro server. Analog phones through our TDM400P do sound much better but the audio problems on our Cisco SIP phones are echo problems. People are working on solutions. Tom >Cheers, > >Brian >___ >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] Asterisk and Cisco 7960 problems persist (for me, anyway)
> It seems that each time I get a new checkout of * from CVS my Cisco 7960 > works worse than before. I know this stuff's in flux, so I mention this > in case it's news. Anyone else having trouble? What I'm seeing (er, > hearing) is really choppy audio. The previous version I had installed > had fairly frequent audio dropouts (not present when I make the same > calls through the same * box using a TDM400P interface). Brian, Are you having the choppy audio only on iax2 links or on other calls as well? There was an issue with erratic iax2 timestamps which caused the Cisco phones to effectively drop any sip packet that had uneven timestamps causing extremely choppy audio. The choppy audio (as I seen it) was only in one direction (from the iax2 source with the erratic timestamps towards to 7960 phone). If your issue is not associated with iax2, then be aware the Cisco v6.x code changed DSP firmware internally, and any sip/rtp packets arriving with uneven timestamps (within the rtp pkts) will be dropped and cause the choppy audio. You should be able to see the timestamps with ethereal. The timestamp difference between successive pkts should be exactly 160 milliseconds; if its anything else, the phone will drop the pkt. (Not sure if that is a real Cisco bug or a planned change, but it certainly has a hugh negative impact on voice quality.) I'm running CVS-HEAD-05/02/04 with no problems today. 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
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
At 09:43 AM 5/7/2004, you wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). No dropout problems or choppy audio running Asterisk CVS-04/19/04-14:31:03 with 4 Cisco 7940/60 SIP 6.3 phones on a 2.4GHz P4 Supermicro server. Analog phones through our TDM400P do sound much better but the audio problems on our Cisco SIP phones are echo problems. People are working on solutions. Tom Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
It's not the switch. It's lightly loaded 100Mb. -brian Bisker, Scott (7805) wrote: What kind of switch do you have your phones plugged into? If your switch is highly loaded, or you are doing lots of multicast or broadcast, your SIP streams are going to suffer unless you are filtering that traffic at the port level or have separate VOIP VLANS. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie Sent: Friday, May 07, 2004 10:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
I've had this too, reported it as a bug last week and got my butt kicked for not being responsive enough in providing support to sort it out. You could file another bug report but be sure to have a thick book ready to stuff down your trousers. Iain --On Friday, May 7, 2004 10:43 am -0400 Brian Cuthie <[EMAIL PROTECTED]> wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
What kind of switch do you have your phones plugged into? If your switch is highly loaded, or you are doing lots of multicast or broadcast, your SIP streams are going to suffer unless you are filtering that traffic at the port level or have separate VOIP VLANS. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie Sent: Friday, May 07, 2004 10:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway) It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960 problems persist (for me, anyway)
It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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] Asterisk and Cisco 7960
thanks very much... do you know of any other links to documentation, guides, manuals etc. (Digium site does not offer much). The biggest problem so far, I find is lack of docs. To produce information one does need data. Senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: 05 September 2003 20:27 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and Cisco 7960 On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote: > hi, what does "tr" means at the end of line? There is documentation, it is even within quick access. >From issueing a "show application dial" at a asterisk cli prompt I see the following. The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then picked up by another user. The optionnal URL will be sent to the called party if the channel supports it. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Andrew > Gillham > Sent: 05 September 2003 06:29 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 > > > Andrew Joakimsen wrote: > > >>>exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > >>> > >>> > >>This didn't work - what does the @1000 indicate? > >> > >> > > > > > >It shouldn't be there, If it's defined as 1000 in sip.conf make your > >dial string > > > >exten => 1000,1,Dial(SIP/1000,20,Ttr) > > > > You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are > calling! > > This just says I am calling the line configured as '1000' on the Cisco > device that is defined as [1000] in sip.conf. > > -Andrew > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote: > hi, what does "tr" means at the end of line? There is documentation, it is even within quick access. >From issueing a "show application dial" at a asterisk cli prompt I see the following. The option string may contain zero or more of the following characters: 't' -- allow the called user transfer the calling user 'T' -- to allow the calling user to transfer the call. 'r' -- indicate ringing to the calling party, pass no audio until answered. 'm' -- provide hold music to the calling party until answered. 'd' -- data-quality (modem) call (minimum delay). 'c' -- clear-channel data call (PRI-PRI only). 'H' -- allow caller to hang up by hitting *. 'C' -- reset call detail record for this call. 'P[(x)]' -- privacy mode, using 'x' as database if provided. In addition to transferring the call, a call may be parked and then picked up by another user. The optionnal URL will be sent to the called party if the channel supports it. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Andrew > Gillham > Sent: 05 September 2003 06:29 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 > > > Andrew Joakimsen wrote: > > >>>exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > >>> > >>> > >>This didn't work - what does the @1000 indicate? > >> > >> > > > > > >It shouldn't be there, If it's defined as 1000 in sip.conf make your > >dial string > > > >exten => 1000,1,Dial(SIP/1000,20,Ttr) > > > > You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are > calling! > > This just says I am calling the line configured as '1000' on the Cisco > device that is defined as [1000] in sip.conf. > > -Andrew > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
hi, what does "tr" means at the end of line? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Gillham Sent: 05 September 2003 06:29 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 Andrew Joakimsen wrote: >>>exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) >>> >>> >>This didn't work - what does the @1000 indicate? >> >> > > >It shouldn't be there, If it's defined as 1000 in sip.conf make your >dial string > >exten => 1000,1,Dial(SIP/1000,20,Ttr) > You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are calling! This just says I am calling the line configured as '1000' on the Cisco device that is defined as [1000] in sip.conf. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew Joakimsen wrote: exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? It shouldn't be there, If it's defined as 1000 in sip.conf make your dial string exten => 1000,1,Dial(SIP/1000,20,Ttr) You need 'SIP/[EMAIL PROTECTED]' if you want to tell the Cisco what line you are calling! This just says I am calling the line configured as '1000' on the Cisco device that is defined as [1000] in sip.conf. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Sorry for the late reply on this.. Ben Wern wrote: Andrew, Thanks for your help! I did have the outgoing proxy set -- since I had FWD set up on line 1. I removed all the FWD stuff, and the outgoing proxy. I altered the entry to have the qualify, canreinvite, and nat lines and also altered the user id to be a number. Now I'm able to call other local extensions, but I can't call into the Cisco. But it's progress! The outgoing proxy apparently overrides the per line proxy, so you want to leave it empty and just configure each line with the appropriate proxy. I can also call out to FWD, but audio drops after a few seconds. Don't even want to think about getting FWD calls back into the network. exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? If your sip.conf entry is under '[1000]' and you have the Cisco configured as '1000', this is tell the 7960 which line is ringing. That is why you get the 404 not found because it doesn't know which line you're calling. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Ben Wern > Sent: Saturday, August 30, 2003 3:02 AM > To: [EMAIL PROTECTED]; [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960 > > Andrew, > > Thanks for your help! No problem, that's what the list is for :) > I did have the outgoing proxy set -- since I had FWD set up on line 1. I > removed all the FWD stuff, and the outgoing proxy. I altered the entry to > have the qualify, canreinvite, and nat lines and also altered the user id > to be a number. Now I'm able to call other local extensions, but I can't > call into the Cisco. But it's progress! > > I can also call out to FWD, but audio drops after a few seconds. Don't > even > want to think about getting FWD calls back into the network. Change your dial strings end in ,Tt) or ,Ttr) > >exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > This didn't work - what does the @1000 indicate? It shouldn't be there, If it's defined as 1000 in sip.conf make your dial string exten => 1000,1,Dial(SIP/1000,20,Ttr) I don't know what the Tt does (lack of documentation) but adding an r has asterisk generate the ringing (when dialing calls to outside providers/cards) and m will insert music on hold in most cases. BEGIN:VCARD VERSION:2.1 N:Joakimsen;Andrew FN:Andrew Joakimsen ([EMAIL PROTECTED]) ORG:Envision Studio TEL;WORK;VOICE:(888) 210-8063 TEL;CELL;VOICE:(305) 776-0334 TEL;WORK;FAX:(305) 669-6720 EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20030819T050904Z END:VCARD
Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, Thanks for your help! I did have the outgoing proxy set -- since I had FWD set up on line 1. I removed all the FWD stuff, and the outgoing proxy. I altered the entry to have the qualify, canreinvite, and nat lines and also altered the user id to be a number. Now I'm able to call other local extensions, but I can't call into the Cisco. But it's progress! I can also call out to FWD, but audio drops after a few seconds. Don't even want to think about getting FWD calls back into the network. exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) This didn't work - what does the @1000 indicate? Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
Andrew, I removed that entry, still no luck. I also altered the config to use a number (101) as the entry name instead. I get: Got SIP response 404 "Not Found" back from 172.16.1.28 SIP/101-e9a4 is circuit-busy on the console when I try to call it. sip show peers shows the node, with an OK status. Ben At 02:09 AM 8/30/2003 -0400, Andrew Joakimsen wrote: > My sip.conf entry for the cisco looks like this: > > [cisco] > type=friend > username=cisco > secret=1234 > host=dynamic > defaultip=[The IP of the 7960] > mailbox= > context=sip > callerid="Ben" <1> Try to remove the defaultip= string. Do you get any errors in the console when it is run in verbose mode? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Cisco 7960
> My sip.conf entry for the cisco looks like this: > > [cisco] > type=friend > username=cisco > secret=1234 > host=dynamic > defaultip=[The IP of the 7960] > mailbox= > context=sip > callerid="Ben" <1> Try to remove the defaultip= string. Do you get any errors in the console when it is run in verbose mode? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
Ben Wern wrote: I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> Use something like: [1000] callerid="Ben" <1000> context=sip type=friend secret=1234 host=dynamic defaultip=youraddress mailbox=1000 Optionally: qualify=500 canreinvite=no nat=yes And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) You might want this to be: exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Make sure you didn't set the 'outbound proxy' setting on the phone, that will force everything to the proxy. Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 Set all the 'cisco' entries to '1000' in this case. I have several 7960s working with Asterisk, so I can help you out more if you need it. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 The FWD line works, the Asterisk line doesn't. Any suggestions or pointers to documentation I might have missed? Thanks, Ben Wern Hints to make your life easier and maybe solve your problem: 1) Make the username numeric instead of "cisco". Change all appropriate configs. 2) Do you see the Cisco trying to register with Asterisk? Use "tethereal port 5060" to watch what happens. 3) Just for fun, put "nat=1" in the peer for your Cisco. It won't hurt. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no luck. I'm sure I'm missing something very easy... since I know others have this working. I've stepped through Andy Powell's excellent "Getting Started with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for the cisco looks like this: [cisco] type=friend username=cisco secret=1234 host=dynamic defaultip=[The IP of the 7960] mailbox= context=sip callerid="Ben" <1> And the related extensions.conf entry: exten => 1,1,Dial(SIP/cisco,20,tr) The Cisco config itself.. Line 1 is set for FWD. Line 2 is: Name: cisco Shortname: cisco Authentication Name: cisco Authentication Password: 1234 Display Name: cisco proxy address: [The IP of my Asterisk installation] proxy port: 5060 The FWD line works, the Asterisk line doesn't. Any suggestions or pointers to documentation I might have missed? Thanks, Ben Wern ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users