I have a softphone I'm trying on a blackberry, that registers on my
Asterisk, can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the phone
registers), and sip show peers confirms it is "unreachable".
Any suggestions? Is this just a dumb client or do I need to tweak an asterisk
setting?
Thanks
pbx*CLI> sip debug peer 230bb
Unable to get IP address of peer '230bb'
The 'sip debug' command is deprecated and will be removed in a future
release. Please use 'sip set debug' instead.
pbx*CLI>
-- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800
[2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: Peer
'230bb' is now UNREACHABLE! Last qualify: 0
pbx*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
230bb/bob 172.31.254.53 D 9653 UNREACHABLE
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Jeff LaCoursiere
2011-Dec-29 04:21 UTC
[asterisk-users] Client - registers but unreachable
On Wed, 2011-12-28 at 23:16 -0500, Michelle Dupuis wrote:> I have a softphone I'm trying on a blackberry, that registers on my > Asterisk, can make outgoing calls, but can't receive calls. > > There is very little traffic with this phone (see debug below - as the > phone registers), and sip show peers confirms it is "unreachable". > > Any suggestions? Is this just a dumb client or do I need to tweak an > asterisk setting? > > Thanks > > pbx*CLI> sip debug peer 230bb > Unable to get IP address of peer '230bb' > The 'sip debug' command is deprecated and will be removed in a future > release. Please use 'sip set debug' instead. > pbx*CLI> > -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 > [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 > sip_poke_noanswer: Peer '230bb' is now UNREACHABLE! Last qualify: 0 > > pbx*CLI> sip show peers > Name/username Host Dyn Nat ACL Port > Status > 230bb/bob 172.31.254.53 D 9653 > UNREACHABLE > --172.31.254.53 is an RFC1918 address. You need to enable NAT on this client. j
Hello, try to configure "keep alive" option on Softphone if there is. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111228/264fa38f/attachment.htm>
Hello, Your blackberry sip client, works in your wifi network? or by blackberry internet? do you set nat=yes if your phone, register by internet? What is your sip.conf? Regards On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote:> I have a softphone I'm trying on a blackberry, that registers on my > Asterisk, can make outgoing calls, but can't receive calls. > > There is very little traffic with this phone (see debug below - as the > phone registers), and sip show peers confirms it is "unreachable". > > Any suggestions? Is this just a dumb client or do I need to tweak an > asterisk setting? > > Thanks > > pbx*CLI> sip debug peer 230bb > Unable to get IP address of peer '230bb' > The 'sip debug' command is deprecated and will be removed in a future > release. Please use 'sip set debug' instead. > pbx*CLI> > -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 > [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: > Peer '230bb' is now UNREACHABLE! Last qualify: 0 > > pbx*CLI> sip show peers > Name/username Host Dyn Nat ACL Port > Status > 230bb/bob 172.31.254.53 D 9653 > UNREACHABLE > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111228/39976966/attachment-0001.htm>
The BB is using wifi, on the same subnet as the asterisk server so no need for NAT. There is no keep alive option on the softphone (very simplistic settings) Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111228/83219e5c/attachment.htm>
You could also set qualify to no for this extension, or increase the qualify timeout. Does the bb have a reliable signal, and consistent connectivity to the network? When you say the phone is on the same wifi network, do you mean that it's actually the same subnet? Or is the asterisk server on a different internal network. For example, maybe your wireless devices are in a DMZ net and the asterisk server is on another network? If so, you may still need to enable NAT for the extension. You should be able to confirm whether or not this is a NAT problem with tshark or tcpdump on the asterisk server. It will be clear what IP the asterisk server thinks it's talking to in the packet trace. On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote:> I have a softphone I'm trying on a blackberry, that registers on my > Asterisk, can make outgoing calls, but can't receive calls. > > There is very little traffic with this phone (see debug below - as the > phone registers), and sip show peers confirms it is "unreachable". > > Any suggestions? Is this just a dumb client or do I need to tweak an > asterisk setting? > > Thanks > > pbx*CLI> sip debug peer 230bb > Unable to get IP address of peer '230bb' > The 'sip debug' command is deprecated and will be removed in a future > release. Please use 'sip set debug' instead. > pbx*CLI> > -- Registered SIP '230bb' at 172.31.254.53 port 9653 expires 1800 > [2011-12-28 23:11:09] NOTICE[9635]: chan_sip.c:15851 sip_poke_noanswer: > Peer '230bb' is now UNREACHABLE! Last qualify: 0 > > pbx*CLI> sip show peers > Name/username Host Dyn Nat ACL Port > Status > 230bb/bob 172.31.254.53 D 9653 > UNREACHABLE > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120101/4a2ff91e/attachment.htm>