 ohmer
join:2003-08-06
Quebec, QC
 ·voip.ms
·Primus Talkbroad..
 ·TekSavvy DSL
4 edits | [Other] voip.ms & pap2t : calls don't always cut when I hang up Hello,
I don't know if it's the voip.ms fault or the pap2t fault. Sometime, when I hangup, the call doesn't seem to disconnect.
Example : I call my cell phone w/o respond and I hangup my voip phone. My cell phone continue to ring. During this time, if I try to call another number from my voip phone, I get reorder (fast busy) and the call doesn't complete. When my cell phone stop ringing, I can complete anothers calls again.
An idea what happen?
I'm using premium routing. |
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PX EliezerPremium
join:2008-08-09
HuttRiver US
kudos:11 Reviews:
·callwithus
·voip.ms
·Vitelity VOIP
·Callcentric
·Optimum Voice
·Gizmo5
| Re: [Other] voip.ms & pap2t : calls don't always cut when I hang Our colleague mango notes:
"You may want to set the CPC delay to 10 and the CPC duration to 0.5. With the default settings, our phones had to be on the hook for an inordinate amount of time before it would actually end the call."
»www.toao.net/25-linksys-pap2t-vo···r-review |
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ohmer
join:2003-08-06
Quebec, QC | Thanks! I applied the settings (and other I found on that page that seem usefull). I will test if this work better. |
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ohmer
join:2003-08-06
Quebec, QC | Crap, still the same thing :/ |
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PX EliezerPremium
join:2008-08-09
HuttRiver US
kudos:11 Reviews:
·callwithus
·voip.ms
·Vitelity VOIP
·Callcentric
·Optimum Voice
·Gizmo5
| Mmmph.
If you haven't already tried them, go to the PAP2T's default settings for these parameters:
CPC Delay 2
CPC Duration 0 (zero)
Those work for me for Voip.MS
If that does not work either, hopefully someone else will have ideas. |
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| reply to ohmer
CPC delay/duration have absolutely nothing to do with what you're seeing. Those values set when and for how long the TA drops the line power when the other end terminates the call. They would be useful if, for example, you had an answering machine that didn't disconnect when the caller did.
Turn on syslog or sniff the packets with wireshark and make sure your TA is sending a SIP CANCEL (if the call is unanswered) or SIP BYE (if answered) to your provider. Syslog is preferred because you should be able to see if the TA thinks you've hung up the phone, but you need a syslog server to receive the messages. Either way, the TA should get back a 200 OK and then a 487 REQUEST CANCELED for a CANCEL, or just a 200 OK for a BYE. If not, you're going to need to contact your service provider. |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| said by trekologer:CPC delay/duration have absolutely nothing to do with what you're seeing. Those values set when and for how long the TA drops the line power when the other end terminates the call. They would be useful if, for example, you had an answering machine that didn't disconnect when the caller did. Turn on syslog or sniff the packets with wireshark and make sure your TA is sending a SIP CANCEL (if the call is unanswered) or SIP BYE (if answered) to your provider. Syslog is preferred because you should be able to see if the TA thinks you've hung up the phone, but you need a syslog server to receive the messages. Either way, the TA should get back a 200 OK and then a 487 REQUEST CANCELED for a CANCEL, or just a 200 OK for a BYE. If not, you're going to need to contact your service provider. Hello,
I enabled syslog (easy because I use Linux so I only had to activate networking on my syslog server). But I found the ATA to not be very verbose... should I put my syslog's IP into "debug server" too? |
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| Yes, you need to put it in both fields and set the debug level to 99. |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| Thanks!
I don't see any BYE or CANCELLED request.
quote:
First call and hang up
Apr 16 07:53:42 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Reg Addr Change(0) 0:5060->ae893fca:5060
Apr 16 07:54:02 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Off Hook
Apr 16 07:54:04 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit * (1)(40 ms)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 7 (1)(40 ms)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 5 (1)(40 ms)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 9 (1)(40 ms)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 6 (1)(40 ms)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX Calling:*7596@sip.ca2.voip.ms:0
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD ALLOC CALL (port=16446)
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Rx Up
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:05 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]ENC INIT 0
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Tx Up (pt=0->ae893fca:16006)
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTCP Tx Up
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX CC:CallProgress
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Rx 1st PKT @16446(2)
Apr 16 07:54:11 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]DEC INIT 0
Apr 16 07:54:14 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: getting alternate from domain:sip.ca2.voip.ms
Apr 16 07:54:14 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX [0]On Hook
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD Rel Call
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX RSE:GetServerAddrErr(sip.ca2.voip.ms,0)=-101
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX TP:?Tx->0
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX DLG Terminated
Apr 16 07:54:22 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Second call. Canot be complete because the previous didn't disconnected.
Apr 16 07:54:24 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Off Hook
Apr 16 07:54:25 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit * (1)(40 ms)
Apr 16 07:54:26 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 7 (1)(40 ms)
Apr 16 07:54:26 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 5 (1)(40 ms)
Apr 16 07:54:26 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 9 (1)(40 ms)
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 6 (1)(40 ms)
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX Calling:*7596@sip.ca2.voip.ms:0
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD ALLOC CALL (port=16448)
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Rx Up
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX RSE:GetServerAddrErr(sip.ca2.voip.ms,0)=-101
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX TP:?Tx->0
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD Rel Call
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX CC:Failed w/ Calling
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX Sess Terminated
Apr 16 07:54:27 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX RSE:GetServerAddrErr(sip.ca2.voip.ms,0)=-101
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Reg Addr Change(0) ae893fca:5060->0:5060
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Reg Addr Change(0) ae893fca:5060->0:5060
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX TP:?Tx->0
Apr 16 07:54:29 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Apr 16 07:54:31 pap2t.lan.xxxx.ca 00259C6XXXXX [0]On Hook
Apr 16 07:54:32 pap2t.lan.xxxx.ca 00259C6XXXXX Discard RSP w/o trnsac
I can now complete anothers calls
Apr 16 07:54:33 pap2t.lan.xxxx.ca 00259C6XXXXX Discard RSP w/o trnsac
Apr 16 07:54:37 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Off Hook
Apr 16 07:54:37 pap2t.lan.xxxx.ca 00259C6XXXXX Sess Terminated
Apr 16 07:54:37 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
Apr 16 07:54:37 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: last unref for domain sip.ca2.voip.ms
Apr 16 07:54:39 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit * (1)(40 ms)
Apr 16 07:54:39 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 7 (1)(40 ms)
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 5 (1)(40 ms)
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 9 (1)(40 ms)
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX 2. Report digit 6 (1)(40 ms)
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX Calling:*7596@sip.ca2.voip.ms:0
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD ALLOC CALL (port=16450)
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Rx Up
Apr 16 07:54:40 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:44 pap2t.lan.xxxx.ca last message repeated 2 times
Apr 16 07:54:44 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Reg Addr Change(0) 0:5060->ae893fca:5060
Apr 16 07:54:44 pap2t.lan.xxxx.ca 00259C6XXXXX [0]Reg Addr Change(0) 0:5060->ae893fca:5060
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]ENC INIT 0
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Tx Up (pt=0->ae893fca:15130)
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTCP Tx Up
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX CC:CallProgress
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]RTP Rx 1st PKT @16450(2)
Apr 16 07:54:47 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]DEC INIT 0
Apr 16 07:54:50 pap2t.lan.xxxx.ca 00259C6XXXXX [0]On Hook
Apr 16 07:54:50 pap2t.lan.xxxx.ca 00259C6XXXXX [0:0]AUD Rel Call
Apr 16 07:54:50 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: reference domain:sip.ca2.voip.ms
Apr 16 07:54:50 pap2t.lan.xxxx.ca 00259C6XXXXX DLG Terminated
Apr 16 07:54:55 pap2t.lan.xxxx.ca 00259C6XXXXX RSE_DEBUG: unref domain, sip.ca2.voip.ms
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| Its been a while since I've debugged a PAP2 (and you don't have SIP debugging on--did you set the syslog server in both the syslog and debug fields and set the level to 99? Possibly there is a line-specific setting as well) but...
When you hang up the first call the TA reports that the port 0 (you start counting from 0 so that's the first phone port, the second one would be port 1) went on hook and call 0 on port 0 was released. But there may be an error when sending the CANCEL to your provider's proxy. |
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ohmer
join:2003-08-06
Quebec, QC | Yes I configured syslog server, debug server and debug level to 99. I will look if I can find a sip debuging option this evening. |
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| reply to ohmer
On Line 1 tab, set SIP Debug Option to full. |
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| reply to ohmer
On the Regional tab, what is "Hook Flash Timer Max" set to? Default is .9. |
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| reply to ohmer
I took a closer look at your debug log; this appears to be a DNS issue. I believe that "getting alternate" during the first call was a failing DNS lookup and the GetServerAddrErr after going on-hook showed that it didn't know where to send the cancel.
Check if you have the latest firmware (I believe it's 5.1.6) -- this unit is notorious for DNS issues.
You might try setting DNS servers to e.g. 4.2.2.1 and 4.2.2.2 and set DNS Server Order to Manual.
Or, try taking DNS out of the picture altogether, by setting an Outbound Proxy of 174.137.63.202 , Use Outbound Proxy to yes, and Use OB Proxy In Dialog to yes. |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| reply to Stewart
said by Stewart:On Line 1 tab, set SIP Debug Option to full. Thanks I see more log now. I will look at this! |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| reply to TBaritone
said by TBaritone:On the Regional tab, what is "Hook Flash Timer Max" set to? Default is .9. It's set to the default, .9. |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| reply to Stewart
said by Stewart:I took a closer look at your debug log; this appears to be a DNS issue. I believe that "getting alternate" during the first call was a failing DNS lookup and the GetServerAddrErr after going on-hook showed that it didn't know where to send the cancel. Check if you have the latest firmware (I believe it's 5.1.6) -- this unit is notorious for DNS issues. You might try setting DNS servers to e.g. 4.2.2.1 and 4.2.2.2 and set DNS Server Order to Manual. Or, try taking DNS out of the picture altogether, by setting an Outbound Proxy of 174.137.63.202 , Use Outbound Proxy to yes, and Use OB Proxy In Dialog to yes. I already upgraded my firmware to the last versiuon (5.1.6).
Changing dns change nothing. |
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| said by ohmer:Changing dns change nothing. Try the outbound proxy with numeric IP, just as a test to see where the problem lies. I agree that it's not a good long-term solution, because if voip.ms changes the address your system would stop working until you make the corresponding change manually.
It would be useful if you can capture traffic from/to the unit, in addition to the syslog. If the issue does prove to be DNS related, you'll be able to see DNS lookups and replies. If not, at least you'll see what network activity (if any) is associated with the cryptic log messages. |
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ohmer
join:2003-08-06
Quebec, QC Reviews:
·voip.ms
·Primus Talkbroad..
·TekSavvy DSL
| said by Stewart:said by ohmer:Changing dns change nothing. Try the outbound proxy with numeric IP, just as a test to see where the problem lies. I agree that it's not a good long-term solution, because if voip.ms changes the address your system would stop working until you make the corresponding change manually. It would be useful if you can capture traffic from/to the unit, in addition to the syslog. If the issue does prove to be DNS related, you'll be able to see DNS lookups and replies. If not, at least you'll see what network activity (if any) is associated with the cryptic log messages. Replacing dns name by ip seem to works... no more dns message in the log.
But replacing ca1/ca2 servers by sip.us1.voip.ms also work (but message apear again in the logs).
The CA servers doesn't like me... |
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| said by ohmer:The CA servers doesn't like me... Be sure that Use DNS SRV is off (I don't believe voip.ms has any SRV records), try setting DNS Query Mode to Sequential, be sure that DNS Server Order is Manual.
If no luck, I recommend capturing traffic to see what is going wrong. IMO there are two likely possibilities: either the DNS lookup is somehow bad, e.g. because of DNS proxying in your network or at your ISP, or the Linksys is interpreting it incorrectly, which might be worked around by a settings change. |
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