Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean
Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hi all,****
** **
First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend.****
** **
I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime.****
** **
Helpful tidbits:****
**· **I THINK it happens in either direction.****
**· **For this person at his home, it appears to be every time (for now)****
**· **He reports calling (to or from) other Sprint Cell users results in the same problem from our FS****
**· **It appears to only be true with Sprint Cell calls! (But my users say that’s not Sprints fault!)****
** **
Scenario:****
I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear … he can’t hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell.****
** **
Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** **
And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip ****
** **
Based on the small size of the file, I suspect someone is going to say “do it again with this tcpdump command”. I welcome the education.****
** **
Anyway, any insight will be appreciated.****
** **
Sean****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number> Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed.*** *
** **
My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number>****
Of course I got 2 way audio. I called the one that ALWAYS fails …. Got 2 way audio! Does pcapsipdump fix it? lol****
** **
I will try in day time tomorrow and see if we can get a failure.****
** **
Nothing in the freeswitch.log of value? I didn’t see anything, but there is still a lot for me to learn there.****
** **
Sean****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, May 16, 2013 7:35 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] One Way Audio****
** **
Sean,****
Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options:****
Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question.****
Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it.****
My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it.****
-MC****
** **
On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: ****
****
Hi all,****
****
First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend.****
****
I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime.****
****
Helpful tidbits:****
· I THINK it happens in either direction.****
· For this person at his home, it appears to be every time (for now)****
· He reports calling (to or from) other Sprint Cell users results in the same problem from our FS****
· It appears to only be true with Sprint Cell calls! (But my users say that’s not Sprints fault!)****
****
Scenario:****
I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear … he can’t hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell.****
****
Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** **
And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip ****
****
Based on the small size of the file, I suspect someone is going to say “do it again with this tcpdump command”. I welcome the education.****
****
Anyway, any insight will be appreciated.****
****
Sean****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Thanks MC. Let's shelve this for now. It appears to have been specific to one of our gateways. If it returns I will throw it at you. Thanks again. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number> Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number> Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hey Michael,****
** **
The problem is back and seems to all the time now.****
** **
Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz ****
** **
I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. ****
** **
Also, any tips on getting started with wireshark to investigate myself next time would be appreciated.****
** **
Thanks,****
Sean****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Friday, May 17, 2013 7:29 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] One Way Audio****
** **
It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching...****
-MC****
** **
On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: ****
Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed.*** *
****
My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number>****
Of course I got 2 way audio. I called the one that ALWAYS fails …. Got 2 way audio! Does pcapsipdump fix it? lol****
****
I will try in day time tomorrow and see if we can get a failure.****
****
Nothing in the freeswitch.log of value? I didn’t see anything, but there is still a lot for me to learn there.****
****
Sean****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, May 16, 2013 7:35 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] One Way Audio****
****
Sean,****
Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options:****
Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question.****
Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it.****
My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it.****
-MC****
****
On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: ****
****
Hi all,****
****
First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend.****
****
I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime.****
****
Helpful tidbits:****
· I THINK it happens in either direction.****
· For this person at his home, it appears to be every time (for now)****
· He reports calling (to or from) other Sprint Cell users results in the same problem from our FS****
· It appears to only be true with Sprint Cell calls! (But my users say that’s not Sprints fault!)****
****
Scenario:****
I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear … he can’t hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell.****
****
Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** **
And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip ****
****
Based on the small size of the file, I suspect someone is going to say “do it again with this tcpdump command”. I welcome the education.****
****
Anyway, any insight will be appreciated.****
****
Sean****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Hey Michael, Thanks. I am just throwing this out here for anyone who may stumble on this in archives. My first thought was you just punted! However, I installed wireshark and had a look. I saw enough to figure out what your statement "You are definitely sending RTP" means! FS is send RTP (Real Time Protocol) packets to the carrier. The RTP packets ARE the audio (profound revelation) . So everything is fine with FS and the audio is getting lost between my Carrier (Vitelity) and SPRINT. It is odd that we have had no complaints of lost audio except to SPRINT phones. And I must say I do not have this problem on my SPRINT phone except for calls from my FS. I will post Vitelity's response. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 22, 2013 3:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number> Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com> wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
On Wed, May 22, 2013 at 8:07 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Hey Michael,****
** **
Thanks.****
** **
I am just throwing this out here for anyone who may stumble on this in archives.
Thanks for sharing your journey along the way. Every new user has this journey and there are occasionally some bumps. You were wise to roll up your sleeves and dive in. That's the best way to learn. While all of this stuff is documented on the wiki and elsewhere on the Web there is simply no substitute for seeing it firsthand. Wireshark is great for that.
****
** **
My first thought was you just punted! However, I installed wireshark and had a look. I saw enough to figure out what your statement “You are definitely sending RTP” means! FS is send RTP (Real Time Protocol) packets to the carrier. The RTP packets ARE the audio (profound revelation) . So everything is fine with FS and the audio is getting lost between my Carrier (Vitelity) and SPRINT.
Sometimes the simplest truths are the most profound. ("Is the world round or flat?" or "Is my wife pregnant or not?" :)
****
** **
It is odd that we have had no complaints of lost audio except to SPRINT phones. And I must say I do not have this problem on my SPRINT phone except for calls from my FS.****
** **
I will post Vitelity’s response.
We are eager to hear what they have to say. -MC
****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Wednesday, May 22, 2013 3:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] One Way Audio****
** **
You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP.****
-MC****
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Try limiting the codec to just g711u for testing, I remember having something similar with another solution also with Sprint/vitelity before I started using Freeswitch, I don't remember the resolution however I think it was Codec related. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Wednesday, May 22, 2013 11:07 AM To: 'FreeSWITCH Users Help' Subject: spam>spam>Re: [Freeswitch-users] One Way Audio Hey Michael, Thanks. I am just throwing this out here for anyone who may stumble on this in archives. My first thought was you just punted! However, I installed wireshark and had a look. I saw enough to figure out what your statement "You are definitely sending RTP" means! FS is send RTP (Real Time Protocol) packets to the carrier. The RTP packets ARE the audio (profound revelation) . So everything is fine with FS and the audio is getting lost between my Carrier (Vitelity) and SPRINT. It is odd that we have had no complaints of lost audio except to SPRINT phones. And I must say I do not have this problem on my SPRINT phone except for calls from my FS. I will post Vitelity's response. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 22, 2013 3:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n <my number> Of course I got 2 way audio. I called the one that ALWAYS fails .... Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy <sdevoy@bizfocused.com<mailto:sdevoy@bizfocused.com>> wrote: [cid:image003.gif@01CE56E2.8A2DFB70] Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: * I THINK it happens in either direction. * For this person at his home, it appears to be every time (for now) * He reports calling (to or from) other Sprint Cell users results in the same problem from our FS * It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear ... he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
participants (3)
-
Michael Collins
-
Moishe Grunstein
-
Sean Devoy