Re: [asterisk-users] RTP timestamps
Hi, One more interesting fact, i see correlation with DTMF features, after i disabled corresponding options on dial commands (like htw) the timestamps on rtp are constantly growing and no more one way audio problems after call transfer, hold, parking etc. So it seems there is a bug related to rtp, rfc2833 and timestamp calculation. Or maybe some misconfigured features ? Has anyone seen this behaviour before ? Greetings, Liivo 27.10.2009 16:53, Liivo Vöörmann kirjutas: Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP timestamps
Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should be, but sometimes while the asterisk plays MOH (or somebody transfers call to another extension) the timestamps on RTP packets will fall to past. Providers media gateway dosn't like that. The marker bit is correctly set but it seems like that dosn't change anything. Sequences and SSRC-s are Ok, no packet loss. Has anyone seen something like this before and knows what is the cause and how to fix this? I've tried many changes in config and upgraded to 1.6.1 but it didnt change anything, currently running asterisk 1.4.26.1 on 64 bit intel platform with opensuse. Here is the tcpdump view from wireshark, xxx is providers ip and yyy is asterisk: 6218207.717454xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680 6219207.717481yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496 6220207.737442xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54365, Time=1987711840 6221207.757430xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54366, Time=1987712000 6222207.759283yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22827, Time=736089280, Mark 6223207.765349yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22828, Time=736089440 Help! Greetings, Liivo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP timestamps
Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP timestamps
Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP timestamps (was: AArgh, * and the 7960)
On Wed, 19 May 2004, Rich Adamson wrote: As you already know, the timestamp problem with cisco phones dropping packets has been associated with at least two channels. snip Although iax capi are getting hammered, the real issue seems to be that no one has opened a high sev TAC case to fix the root problem. If I was the one to open the TAC case, the transcript would read: VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!! I have not read RFC1889 (RTP) in detail, but I am positive that the timestamp field was put there for a reason. Sure, maybe Cisco is a little overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO. * needs to generate RTP streams with valid timestamp progression -- surely we're not happy to say the Cisco 79x0 is the only phone that cares about timestamps, so there's the problem. (Eek, I'm defending Cisco... Too many beers at lunchtime, maybe.) Right, I'm fed up with looking like a whinger on this list, and will whinge no more. I'm off to try and get to the bottom of this. Maybe I'll stay up until 2am (GMT+10) to try and get kram on IRC; maybe I'll decide that I need the sleep. Cheers, Vic Cross ___ 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] RTP timestamps (was: AArgh, * and the 7960)
As you already know, the timestamp problem with cisco phones dropping packets has been associated with at least two channels. snip Although iax capi are getting hammered, the real issue seems to be that no one has opened a high sev TAC case to fix the root problem. If I was the one to open the TAC case, the transcript would read: VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!! I have not read RFC1889 (RTP) in detail, but I am positive that the timestamp field was put there for a reason. Sure, maybe Cisco is a little overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO. * needs to generate RTP streams with valid timestamp progression -- surely we're not happy to say the Cisco 79x0 is the only phone that cares about timestamps, so there's the problem. (Eek, I'm defending Cisco... Too many beers at lunchtime, maybe.) Right, I'm fed up with looking like a whinger on this list, and will whinge no more. I'm off to try and get to the bottom of this. Maybe I'll stay up until 2am (GMT+10) to try and get kram on IRC; maybe I'll decide that I need the sleep. Vic, You're not the only one with this frustration / show stopper, and regardless of what bkw says, its a serious problem that for whatever reason, a few arrogant people refuse to acknowledge. Problems (just for bkw): 1. Stable does not include the major iax2 fixes that were corrected last month 2. Head has a iax2/gsm sequence number issue both result in choppy audio when transcoded to sip on Cisco phones. As has been noted MULTIPLE TIMES on this list, there is no realistic combination of cvs code that we can use, period. And since at least some of us are not programmers, we're simply body slammed by the arrogant few. Off now to escalate this son of a bitch, or dump asterisk 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] RTP timestamps (was: AArgh, * and the 7960)
Problems (just for bkw): 1. Stable does not include the major iax2 fixes that were corrected last month 2. Head has a iax2/gsm sequence number issue both result in choppy audio when transcoded to sip on Cisco phones. As has been noted MULTIPLE TIMES on this list, there is no realistic combination of cvs code that we can use, period. And since at least some of us are not programmers, we're simply body slammed by the arrogant few. I presented a stopgap solution to this list earlier this week (TDMoE between two * boxes to give you clean RTP timestamps to the Ciscos connected to *1, and *2 to the world) I agree it's not ideal *BUT* it is a solution, and a cheap-as-hell one, too. I'm sure you've got i686-class machine with 64MB and a 2GB HDD around that you can put a couple of ethernet cards in to for the moment. -A. ___ 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] RTP timestamps (was: AArgh, * and the 7960)
You're not the only one with this frustration / show stopper, and regardless of what bkw says, its a serious problem that for whatever reason, a few arrogant people refuse to acknowledge. Get ethereal traces where we can see timestamps and we can figure it out. My phones aren't doing this. Other than that I can't reproduce the problem, trust me I HAVE tried. bkw ___ 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] RTP timestamps (was: AArgh, * and the 7960)
I presented a stopgap solution to this list earlier this week (TDMoE between two * boxes to give you clean RTP timestamps to the Ciscos connected to *1, and *2 to the world) That's just silly to have to do this. Collect me some timestamp info so we can see this invalid timestamp non-sense that's the ONLY way we can fix it. As I have stated every single time it doesn't happen for me or digium and we use 7960's extensively. bkw ___ 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] RTP timestamps (was: AArgh, * and the 7960)
Problems (just for bkw): 1. Stable does not include the major iax2 fixes that were corrected last month 2. Head has a iax2/gsm sequence number issue both result in choppy audio when transcoded to sip on Cisco phones. As has been noted MULTIPLE TIMES on this list, there is no realistic combination of cvs code that we can use, period. And since at least some of us are not programmers, we're simply body slammed by the arrogant few. I presented a stopgap solution to this list earlier this week (TDMoE between two * boxes to give you clean RTP timestamps to the Ciscos connected to *1, and *2 to the world) I agree it's not ideal *BUT* it is a solution, and a cheap-as-hell one, too. I'm sure you've got i686-class machine with 64MB and a 2GB HDD around that you can put a couple of ethernet cards in to for the moment. That's not even close to an acceptable solution. Try this for perspective... A new service provider is getting ready to provide iax termination service to asterisk users throughout the world. Lots of systems and agreements in place. There is no usable cvs code to start the offering. Your approach dictates he install a redundant system at all locations to handle this problem. I don't need a calculator to figure out that does not scale worth a damn. (And, no putting one central system somewhere is not possible.) Problem has now been escalated. ___ 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] RTP timestamps (was: AArgh, * and the 7960)
That's not even close to an acceptable solution. Try this for perspective... A new service provider is getting ready to provide iax termination service to asterisk users throughout the world. Lots of systems and agreements in place. There is no usable cvs code to start the offering. Your approach dictates he install a redundant system at all locations to handle this problem. I don't need a calculator to figure out that does not scale worth a damn. (And, no putting one central system somewhere is not possible.) Problem has now been escalated. If you are having these issues please find me on IRC in #asterisk-bugs have ethereal installed and ready and let me ssh in and look at it, I will do my best to find the problem. bkw ___ 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] RTP timestamps (was: AArgh, * and the 7960)
On Fri, 21 May 2004, brian wrote: Get ethereal traces where we can see timestamps and we can figure it out. I did. See a post a few days ago (in the AArgh thread). But I'm the chan_capi problem guy... :( Other than that I can't reproduce the problem, trust me I HAVE tried. Appreciate it. Vic Cross ___ 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] RTP timestamps (was: AArgh, * and the 7960)
Care to find me on IRC and let me look at it and see if I can fix it? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vic Cross Sent: Friday, May 21, 2004 9:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP timestamps (was: AArgh, * and the 7960) On Fri, 21 May 2004, brian wrote: Get ethereal traces where we can see timestamps and we can figure it out. I did. See a post a few days ago (in the AArgh thread). But I'm the chan_capi problem guy... :( Other than that I can't reproduce the problem, trust me I HAVE tried. Appreciate it. Vic Cross ___ 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] RTP timestamps (was: AArgh, * and the 7960)
It does look like stable can send the wrong timestamps but that wont fix the capi issue as it will have to be updated also. But in my testing it didn't mess with the audio. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Friday, May 21, 2004 9:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP timestamps (was: AArgh, * and the 7960) Care to find me on IRC and let me look at it and see if I can fix it? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vic Cross Sent: Friday, May 21, 2004 9:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP timestamps (was: AArgh, * and the 7960) On Fri, 21 May 2004, brian wrote: Get ethereal traces where we can see timestamps and we can figure it out. I did. See a post a few days ago (in the AArgh thread). But I'm the chan_capi problem guy... :( Other than that I can't reproduce the problem, trust me I HAVE tried. Appreciate it. Vic Cross ___ 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] RTP timestamps (was: AArgh, * and the 7960)
Given what we just went through this morning with iax/gsm, there is something different about how your 7960's deal with the uneven timestamps. Therefore, your tests with capi are not likely to be reflective of Vic's problem either. Wish we know what was different. It does look like stable can send the wrong timestamps but that wont fix the capi issue as it will have to be updated also. But in my testing it didn't mess with the audio. bkw -Original Message- Care to find me on IRC and let me look at it and see if I can fix it? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vic Cross Sent: Friday, May 21, 2004 9:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP timestamps (was: AArgh, * and the 7960) On Fri, 21 May 2004, brian wrote: Get ethereal traces where we can see timestamps and we can figure it out. I did. See a post a few days ago (in the AArgh thread). But I'm the chan_capi problem guy... :( Other than that I can't reproduce the problem, trust me I HAVE tried. Appreciate it. Vic Cross ___ 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] RTP timestamps
This time stamp issue is all gone.. now if everyone will just UPDATE! bkw - Original Message - From: Andres [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 4:14 PM Subject: Re: [Asterisk-Users] RTP timestamps VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!! I have not read RFC1889 (RTP) in detail, but I am positive that the timestamp field was put there for a reason. Sure, maybe Cisco is a little overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO. * needs to generate RTP streams with valid timestamp progression -- surely we're not happy to say the Cisco 79x0 is the only phone that cares about timestamps, so there's the problem. Hi Vic, For your information Sipura also suffers from the Timestamp issue. 3 months ago when I opened the case with them, they explained in detail why they needed those Timestamps (it has to do with the jitter buffer calculation algorithms). They told me the problem had to be solved at the Asterisk side since there is no reason why the Timestamps should change. They have not seen this weird behaviour with any other SIP system besides Asterisk. In any case, thats why we came up with the rtp.c hack, and have been happy ever since. -- Andres Network Admin http://www.telesip.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 ___ 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] RTP timestamps
brian k. west wrote: This time stamp issue is all gone.. now if everyone will just UPDATE! bkw It is not gone. We updated one more time all our Asterisk boxes l0 days ago (fresh checkout). And the problem was also evident (no other providers involved in the tests, just calls between all our boxes). We quickly put in place the rtp.c hack and all was back to normal again. -- Andres Network Admin http://www.telesip.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
Re: [Asterisk-Users] RTP timestamps
On Fri, 2004-05-21 at 18:22, Andres wrote: It is not gone. We updated one more time all our Asterisk boxes l0 days ago (fresh checkout). And the problem was also evident (no other providers involved in the tests, just calls between all our boxes). We quickly put in place the rtp.c hack and all was back to normal again. The fix in CVS -stable was about 10 HOURS ago. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] RTP timestamps
Watch the CVS logs the update was put in stable TODAY for a timestamp issue in chan_iax2.c bkw - Original Message - From: Andres [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 5:22 PM Subject: Re: [Asterisk-Users] RTP timestamps brian k. west wrote: This time stamp issue is all gone.. now if everyone will just UPDATE! bkw It is not gone. We updated one more time all our Asterisk boxes l0 days ago (fresh checkout). And the problem was also evident (no other providers involved in the tests, just calls between all our boxes). We quickly put in place the rtp.c hack and all was back to normal again. -- Andres Network Admin http://www.telesip.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 ___ 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] RTP timestamps
Excellent! I'll give it a shot. -brian brian k. west wrote: This time stamp issue is all gone.. now if everyone will just UPDATE! bkw - Original Message - From: Andres [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 21, 2004 4:14 PM Subject: Re: [Asterisk-Users] RTP timestamps VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!! I have not read RFC1889 (RTP) in detail, but I am positive that the timestamp field was put there for a reason. Sure, maybe Cisco is a little overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO. * needs to generate RTP streams with valid timestamp progression -- surely we're not happy to say the Cisco 79x0 is the only phone that cares about timestamps, so there's the problem. Hi Vic, For your information Sipura also suffers from the Timestamp issue. 3 months ago when I opened the case with them, they explained in detail why they needed those Timestamps (it has to do with the jitter buffer calculation algorithms). They told me the problem had to be solved at the Asterisk side since there is no reason why the Timestamps should change. They have not seen this weird behaviour with any other SIP system besides Asterisk. In any case, thats why we came up with the rtp.c hack, and have been happy ever since. -- Andres Network Admin http://www.telesip.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 ___ 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
