-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I have a server currently running Asterisk 1.0.7 placed out in the wild (i.e. not behind NAT). I have groups of sip clients all behind various NAT firewalls (mainly adsl routers). Up to now I've mainly used Sipuras and not had any serious problems. Recently I've been experimenting with Snom phones and I have encountered problems where the Snoms register fine initially but after a while (which could be anything from 2minutes to 45 minutes) they lose their registration. Sample snom configuration in sip.conf follows: [888120] type=friend username=888120 mailbox=888120 canreinvite=no nat=yes secret=secret host=dynamic qualify=yes context=sipdemo subscribecontext=sipdemo I've experimented with several different adsl routers and was surprised at the difference this can make, however the problem is still there to a greater or lesser extent. I've also tried using a Stun server following recommendation here: http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Again this makes a difference, but doesn't entirely solve the problem - there are still occasions where the Snom is unreachable or unknown. The implication seems to be that if asterisk does not send keepalives often enough then the way through the nat is lost. I've also tried lowering the expiry time of the asterisk sessions (in increments down to 30 seconds) in the hope that it would result in more activity and keep the firewall open, but it didn't help. Another strange factor is using the BLF on snoms - the situation seems to be worse with those enabled, but that might not be relevant. So I guess I have a few questions: 1) Has anyone had this happen before and what, if any, was the solution? 2) How do I increase the frequency with which asterisk sends keepalives? 3) Does SER handle this better - would placing this outside the NAT help handle connections from inside? 4) Do newer versions of asterisk handle this better? 5) Any other suggestions? TIA. - -- Richard Watson -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDeGAzP05lUVhVYk0RAkM1AKCepBdfTkLoqwNlnbMpH3CWGTWCcwCeOFlE jbKdXnKHNqG7951KlctSfek=ttdo -----END PGP SIGNATURE-----
Domenico Lanteri
2005-Nov-14 03:37 UTC
[Asterisk-Users] Attended transfer and group problem
I have a problem with group and attended transfer.
I have tested below example dialplan with asterisk-1.2.0-beta1,
asterisk-1.2.0-rc1 and and astesik-HEAD on 11/14/2005.
I have simple test dialplan like:
[default]
exten => 210,1,Macro(stdexten,${EXTEN},SIP,test1)
exten => 211,1,Macro(stdexten,${EXTEN},SIP,test2)
exten => 212,1,Macro(stdexten,${EXTEN},SIP,test3)
[macro-stdexten]
exten => s,1,Set(OUTBOUND_GROUP=${ARG1})
exten => s,n,Set(GROUP=${CALLERIDNUM})
exten => s,n,Dial(${ARG2}/${ARG3}||t)
If i dial 211 from 210, 211 answer and make attended trasfer to 212, 212
answer and 211 hangup, cli show:
Monitor*CLI> set verbose 4
Verbosity is at least 4
-- Executing Macro("SIP/test1-2f4f",
"stdexten|211|SIP|test2") in new
stack
-- Executing Set("SIP/test1-2f4f", "OUTBOUND_GROUP=211")
in new stack
-- Executing Set("SIP/test1-2f4f", "GROUP=210") in new
stack
-- Executing NoOp("SIP/test1-2f4f", "EXTEN: -211-
CALLERIDNUM: -210-")
in new stack
-- Executing NoOp("SIP/test1-2f4f",
"------SIP/test1-2f4f--------") in
new stack
-- Executing Dial("SIP/test1-2f4f", "SIP/test2||t") in
new stack
-- Called test2
-- SIP/test2-c1b6 is ringing
-- SIP/test2-c1b6 answered SIP/test1-2f4f
-- Attempting native bridge of SIP/test1-2f4f and SIP/test2-c1b6
-- Attempting native bridge of SIP/test1-2f4f and SIP/test2-c1b6
-- Started music on hold, class 'default', on channel
'SIP/test1-2f4f'
-- Playing 'pbx-transfer' (language 'it')
-- Executing Macro("Local/212@default-8f24,2",
"stdexten|212|SIP|test3")
in new stack
-- Executing Set("Local/212@default-8f24,2",
"OUTBOUND_GROUP=212") in
new stack
-- Executing Set("Local/212@default-8f24,2",
"GROUP=211") in new stack
-- Executing NoOp("Local/212@default-8f24,2", "EXTEN: -212-
CALLERIDNUM: -211-") in new stack
-- Executing NoOp("Local/212@default-8f24,2",
"------Local/212@default-8f24,2--------") in new stack
-- Executing Dial("Local/212@default-8f24,2",
"SIP/test3||t") in new
stack
-- Called test3
-- SIP/test3-61de is ringing
-- Local/212@default-8f24,1 is ringing
-- SIP/test3-61de answered Local/212@default-8f24,2
Nov 14 12:01:51 NOTICE[6186]: res_features.c:1124
ast_feature_request_and_dial: Don't know what to do about control frame: -1
-- Stopped music on hold on SIP/test1-2f4f
-- Playing 'beep' (language 'en')
== Spawn extension (macro-stdexten, s, 5) exited non-zero on
'Transfered/SIP/test1-2f4f<ZOMBIE>' in macro 'stdexten'
== Spawn extension (default, 211, 1) exited non-zero on
'Transfered/SIP/test1-2f4f<ZOMBIE>'
Monitor*CLI> group show channels
Channel Group Category
SIP/test1-2f4f 210 (default)
SIP/test3-61de 212 (default)
Local/212@default-8f24,2 211 (default)
7 active channels
Channel Local/212 belong to group 211 but the 211 phone is hangup.
Anyone as any idea ?
Thank you
Lanteri Domenico.
Does the phone ocasionally prompt the user for a password? -Mike> -----Original Message----- > From: Richard Watson [mailto:richard@openia.com] > Sent: Monday, November 14, 2005 5:00 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Snom clients deregistering > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi all, > > I have a server currently running Asterisk 1.0.7 placed out > in the wild (i.e. not behind NAT). > > I have groups of sip clients all behind various NAT firewalls > (mainly adsl routers). > > Up to now I've mainly used Sipuras and not had any serious problems. > Recently I've been experimenting with Snom phones and I have > encountered problems where the Snoms register fine initially > but after a while (which could be anything from 2minutes to > 45 minutes) they lose their registration. Sample snom > configuration in sip.conf follows: > > [888120] > type=friend > username=888120 > mailbox=888120 > canreinvite=no > nat=yes > secret=secret > host=dynamic > qualify=yes > context=sipdemo > subscribecontext=sipdemo > > I've experimented with several different adsl routers and was > surprised at the difference this can make, however the > problem is still there to a greater or lesser extent. > > I've also tried using a Stun server following recommendation here: > > http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_aud > io_asterisk.html > > Again this makes a difference, but doesn't entirely solve the > problem - there are still occasions where the Snom is > unreachable or unknown. > > The implication seems to be that if asterisk does not send > keepalives often enough then the way through the nat is lost. > > I've also tried lowering the expiry time of the asterisk > sessions (in increments down to 30 seconds) in the hope that > it would result in more activity and keep the firewall open, > but it didn't help. > > Another strange factor is using the BLF on snoms - the > situation seems to be worse with those enabled, but that > might not be relevant. > > So I guess I have a few questions: > > 1) Has anyone had this happen before and what, if any, was > the solution? > > 2) How do I increase the frequency with which asterisk sends > keepalives? > > 3) Does SER handle this better - would placing this outside > the NAT help handle connections from inside? > > 4) Do newer versions of asterisk handle this better? > > 5) Any other suggestions? > > TIA. > > - -- > Richard Watson > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.1 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org > > iD8DBQFDeGAzP05lUVhVYk0RAkM1AKCepBdfTkLoqwNlnbMpH3CWGTWCcwCeOFlE > jbKdXnKHNqG7951KlctSfek> =ttdo > -----END PGP SIGNATURE----- > >
Michael Crown wrote:> Does the phone ocasionally prompt the user for a password? -MikeYes it does!!!! How did you know?
There is a setting on the "Advanced" page called "Challenge Response on Phone". Turn this setting to "Off" and your problem will be solved. Also, we usually set the "Proposed Expiry" to 1 minute On the "SIP" page when phones are behind a NAT. -Mike> -----Original Message----- > From: Richard Watson [mailto:richard@openia.com] > Sent: Monday, November 14, 2005 8:30 AM > To: mike@thevoipconnection.com; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Snom clients deregistering > > Michael Crown wrote: > > Does the phone ocasionally prompt the user for a password? -Mike > > Yes it does!!!! > > How did you know? >
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 The VoIP Connection wrote:> There is a setting on the "Advanced" page called "Challenge Response on > Phone". Turn this setting to "Off" and your problem will be solved. Also, we > usually set the "Proposed Expiry" to 1 minute On the "SIP" page when phones > are behind a NAT.That doesn't seem to have helped entirely. The "Password" prompt no longer appears but the phone still becomes UNREACHABLE then UNKNOWN after a few minutes. In the system information on the phone it reports "Registration Failed". However a few minutes later it logs itself back in. I have two identical snoms on the bench here and they both do the same thing, logging in and operating fine, before eventually (but not necessarily at the same time) losing registration and stopping for a few minutes. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD4DBQFDeKPnP05lUVhVYk0RAqTfAJYtZqmp1dCRLDhu3C1jHRCeUk5LAJ42z2rV 5Jr8qm+Ruyvv3h2L3jOjUA==PlHs -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Michael Crown wrote:> Did you change the proposed expiry? -MikeYes, now set to 1 minute. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDeLLTP05lUVhVYk0RAuHeAJwOio/yEfblrUEnIaQsjXVbaqdj8gCfQfMC FjPmGjtICurLTdN9DAiXQVg=JgQF -----END PGP SIGNATURE-----
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Richard Watson wrote:> [888120] > type=friend > username=888120 > mailbox=888120 > canreinvite=no > nat=yes > secret=secret > host=dynamic > qualify=yes > context=sipdemo > subscribecontext=sipdemoJust for fun I had a play yesterday using SER as a stateless proxy ouside the nat to see if for some reason that hung on to the registrations. The result was even worse than before. Current situation - I've removed the Challenge Password Dialog configuration from the Snoms and they still lose their registration. I'm quite stumped now - anyone got any idea what to try next? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDea8LP05lUVhVYk0RAkpuAKCAgdcEPxgzqQc9S9jYvHRpQAhWCACcCpuh crOxBqrTfSwp5dtCm9jJGxs=jK4e -----END PGP SIGNATURE-----