[asterisk-users] trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr [EMAIL PROTECTED]:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. sip show peers shows my phones registered OK, but the peer describing my SIP trunk does not even display: sip show peers Name/username HostDyn Nat ACL Port Status 204/204192.168.xxx.xxx D 2048 Unmonitored 203/203192.168.xxx.xxx D 2048 Unmonitored sip show registry sip.voipfone.co.uk:5060 45 Registered Thu, 27 Nov 2008 11:01:56:03 sip reload or restarting asterisk with /etc/init.d/asterisk restart fixes the problem and I get the following output: Name/username HostDyn Nat ACL Port Status 204/204192.168.xxx.xxx D 2048 Unmonitored 203/203192.168.xxx.xxx D 2048 Unmonitored voipfone/ 195.189.173.10 5060 OK (61 ms) sip show registry sip.voipfone.co.uk:5060 45 Registered Thu, 27 Nov 2008 11:05:28:02 sip.conf entry for the trunk [voipfone] type=friend secret=xx username= fromuser= fromdomain=sip.voipfone.co.uk host=sip.voipfone.co.uk insecure=very dtmfmode=rfc2833 context=fromvoipfone ;inbound calls falls in this context of dialplan disallow=all allow=ilbc ;allow=ulaw ;allow=alaw qualify=yes Any ideas warmly welcomed! Setting debug to level 9 isn't helping me out on this. John ___ -- 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] call queuing not detecting caller hang up when call originates from voip provider
Dear all
I've got call queuing working when calls originate from my local site.
After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a phantom until one of the extensions answers it and
it is destroyed
Am I doing something wrong? Am using asterisk 1.4.16.2
Relevant part of files:
sip.conf
[voipfone]
type=friend
secret=
username=xx
fromuser=xx
fromdomain=sip.voipfone.co.uk
host=sip.voipfone.co.uk
insecure=very
dtmfmode=rfc2833
context=fromvoipfone
[s450]
type=friend
context=phones
host=dynamic
[xlite]
type=friend
context=phones
host=dynamic
[consult]
type=friend
context=phones
host=dynamic
extensions.conf
[fromvoipfone]
exten= 1234,1,Dial(SIP/consult,3)
exten= 1234,n,Answer
exten= 1234,n,Ringing
exten= 1234,n,Wait(2)
exten= 1234,n,Background(/var/lib/asterisk/sounds/mhqw)
exten= 1234,n,Queue(myqueue|r)
exten= 1234,n,Hangup
[phones]
exten= 1001,1,Dial(SIP/s450)
exten= 1002,1,Dial(SIP/xlite)
exten= 1003,1,Dial(SIP/consult)
exten= _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,r)
exten= _ZX,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)
exten= _Z,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)
queues.conf
[myqueue]
periodic-announce = mhqw
periodic-announce-frequency = 10
music=default
strategy=ringall
timeout=15
retry=5
wrapuptime=0
maxlen=0
announce-frequency=0
announce-holdtime=no
member = SIP/consult,1
context = phones
Any help appreciated!!
John
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[asterisk-users] multiple sip trunks
I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John ___ -- 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] multiple sip trunks
Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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
Re: [asterisk-users] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf Thanks John 2009/12/11 Martin asteriskl...@callthem.info: On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple registry entries... if it didn't ... it would be just a crippled sip endpoint lets say more ... Asterisk can do whatever you want it to do (within reason and technical boundaries); just code it in or request a feature Martin Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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 ___ -- 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
Re: [asterisk-users] multiple sip trunks
I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? John 2009/12/11 Noah Miller noahisaacmil...@gmail.com: I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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
Re: [asterisk-users] multiple sip trunks
I thought so- the fact the server has 20 different registry entries to 20 different account all at the same ITSP shouldn't matter? Can't see any DDI info in the SIP headers unfortunately :( John 2009/12/14 meetmecall i...@meetmecall.nl The easiest solution to deal with this is to have one context with different extensions for the different numbers and route the incoming calls from there. It should look something like this (not a tested piece of asterisk script, just an example to give the idea). Hope it helps :-) Erik de Wild [all_trunks] exten = 31592123456,1,Goto(trunk1,s,1) exten = 31592123457,1,Goto(trunk1,s,1) exten = 31592123458,1,Goto(trunk1,s,1) exten = 3159212,1,Goto(trunk2,s,1) exten = 31592123334,1,Goto(trunk2,s,1) exten = 31592123335,1,Goto(trunk2,s,1) On 14 dec 2009, at 10:39, Olle E. Johansson wrote: 11 dec 2009 kl. 23.21 skrev John Taylor: I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? This is an effect of the Asterisk architecture. We've had many discussions on how to change it, but right now the peer matching on IP/Port can't separate various instances from each other, since they all have the same IP/port. Asterisk simply goes for the first match, which happens to be the last entry with the IP/port in the sip.conf file. /Olle ___ -- 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 ___ -- 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] caller getting cut off intermittently
I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 ___ -- 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] DNS reload on trunks for outgoing calls
Put the commonly used domain names + appropriate ips into /etc/hosts? John 2010/1/4 Steve Howes steve-li...@geekinter.net: On 4 Jan 2010, at 08:34, Remco Barendse wrote: Is there any fix or workaround for the DNS problem (old standing bug that when the box starts and domain names do not resolve quickly enough from DNS then asterisk stops using the outgoing trunks. I read on the list before that it is considered a huge and unacceptable load for asterisk servers to try and resolve the domain names again after some time but it is rather annoying. I don't know about resources of other people but on my boxes i have some cpu cycles that could be used for that :) I now do nightly restarts of asterisk but it still means that at least for one day calls are flowing through expensive PSTN. If anybody knows of a workaround, would be most welcome Install a resolver locally. ___ -- 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
Re: [asterisk-users] caller getting cut off intermittently
We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf Thanks for any help John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 ___ -- 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] caller getting cut off intermittently
Hi, I've now set dtmfmode=rfc2833 and that seems to have fixed it John 2010/1/7 John Taylor j...@vetsurgeon.org.uk: We're now getting this problem on outgoing calls. I've forced the port to 100FD but still no joy. Anyone any ideas how to debug this- have added verbose to logger.conf Thanks for any help John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] caller getting cut off intermittently
Hi all, I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it John 2010/1/4 John Taylor j...@vetsurgeon.org.uk: I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP --Asterisk server--Natted firewall--extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except occasionally an incoming caller is cut off whilst the called extension stays in the call and can hear a DTMF tone (multimon recognises it as tone D). The asterisk log file shows the call stays active despite the incoming caller being cut off. This has happened to all our extensions at some point (a combination of Snoms and Funkwerks). It happens fairly infrequently, and can happen at any point during a call. The public Lenny server's asterisk config is exactly the same as our LAN Ubuntu asterisk server where we never had this problem. The only difference is that the ITSP trunk is now ulaw rather than ilbc. Can anyone help? Relevant files below (trunk and extension codecs are both ulaw) John example extension in sip.conf: [203] type=friend username=203 secret=xx host=dynamic dtmfmode=inband call-limit=2 qualify=yes nat=yes /var/log/asterisk/messages: [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:1] Set(SIP/301x-09f74a00, oh=0) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:2] NoOp(SIP/301x-09f74a00, 01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:3] GotoIf(SIP/301x-09f74a00, 0?bankhols|200|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [301xx...@fromvoipfone:4] GotoIfTime(SIP/301x-09f74a00, 08:30-18:00|mon-fri|*|*?day|100|1) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Goto (day,100,1) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:1] AGI(SIP/301x-09f74a00, /home/john/phpagi/lookup) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Launched AGI Script /home/john/phpagi/lookup [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- AGI Script /home/john/phpagi/lookup completed, returning 0 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:2] Set(SIP/301x-09f74a00, CALLERID(name)=) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:3] Macro(SIP/301x-09f74a00, monitor|01327xx|in) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:1] Set(SIP/301x-09f74a00, CALLFILENAME=/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@macro-monitor:2] Monitor(SIP/301x-09f74a00, wav|/home/john/asterisk/asterisk_recordings/in-20100104_095856-01295259352|m) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Executing [...@day:4] Dial(SIP/301x-09f74a00, SIP/203SIP/204SIP/206SIP/207SIP/220SIP/221|20|t) in new stack [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 203 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 206 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 207 [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- Called 220 [Jan 4 09:58:56] WARNING[10712] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/206-0a005eb8 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/207-09fe2c98 is ringing [Jan 4 09:58:56] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:57] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/220-09fe7748 is ringing [Jan 4 09:58:58] VERBOSE[10712] logger.c: -- SIP/203-0a001138 is ringing [Jan 4 09:58:59] VERBOSE[10712] logger.c: -- SIP/203-0a001138 answered SIP/301x-09f74a00 -- _ -- 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] forward call back up same trunk to external cell phone problem
Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten = s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten = s,1,Dial(SIP/07123123...@voipfone,20,t) What's wrong?! John -- _ -- 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] forward call back up same trunk to external cell phone problem
Hi- can anyone help with this. I'm really stuck as apparently it should work. Is it a problem with the ITSP, with using the same trunk for both legs of the call etc? John On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote: Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten = s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten = s,1,Dial(SIP/07123123...@voipfone,20,t) What's wrong?! John -- _ -- 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] Problem attended transfer with ilbc
I have an Asterisk server on our LAN that serves our office VOIP phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are ulaw/alaw We use attended transfer extensively. If our trunk is ulaw/alaw they work fine. If the trunk is ilbc we have problems 1- incoming PSTN call routed via voipfone SIP down the trunk to our server 2- our phones ring ok, caller can be answered (e.g. by A) 3- A requests attended transfer to another phone (B) on the LAN- incoming caller put on hold, A can talk to B, B can talk to A 4- A hangs up, B is connected to caller. B can hear caller, but caller cannot hear B. Console output: Asked to transmit frame type 64, while native formats is 0x400 (ilbc)(1024) read/write = 0x40 (slin)(64)/0x400 (ilbc)(1024) Running Asterisk 1.6.2.9 on Ubuntu Karmic- self compiled (do not seem to be able to compile deb source package with ilbc, and deb package does not have ilbc) Any idea what may be happening? John -- _ -- 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
[asterisk-users] Snom phones recommended firmware
We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? Thanks John -- _ -- 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
Re: [asterisk-users] Record() Cmd and My SQL
Why not write the file to /tmp using MixMonitor, then use the command option to trigger an AGI script that will move the data into your database then delete the original file? John On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when doing a load balancing , We cannot confine the recording to a particular asterisk machine ( If we have more than one asterisk machine in the topology ). So a centralized mechanism might be better . So that any machine can access the recording . Regards Mahesh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext David Backeberg Sent: Thursday, September 23, 2010 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI , Is there Any way is there so that I can store my recordings directly to a database rather storing the same to a file . Please, please, please tell us why you would want to do that. -- _ -- 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 -- _ -- 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 -- JA Taylor MA VetMB MRCVS Mansion Hill Veterinary Practice 133-137 Main Road Middleton Cheney OX17 2PP 01295 712110 -- _ -- 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
[asterisk-users] tcpdump auto stats script
Before I reinvent the wheel, I'm looking for a script then when run will - launch tcpdump (or equivalent) on the server and capture all SIP and UDP traffic to an IP address - then, rather than me manually analysing with wireshark, will analyze the cap file and produce stats on jitter, lag, delta etc. Thanks for any help John -- _ -- 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
[asterisk-users] cannot answer incoming calls
Have recently installed some Snom phones into an office. Phones are natted and connect to a 1.4 server on a public IP We can make outgoing calls, but are unable to answer incoming calls. The phone rings, but the call cannot be picked up. Other phones on other sites connected to the server are working perfectly. Looking at the SIP trace it appears the phone transmits: Sent to udp:193.33.xx.xx:5060 at 6/1/2011 11:49:20:868 (849 bytes): SIP/2.0 200 Ok Via: SIP/2.0/UDP 193.33.xx.xx:5060;branch=z9hG4bK6e82052c;rport=5060 From: xx sip:07765000...@sip3.office-voip.com;tag=as1b6fc27c To: sip:x_...@79.123.xx.xx:25380;tag=37gg1zu3wp Call-ID: 1b212085091e98387237125f0ab81...@sip3.office-voip.com CSeq: 102 INVITE Contact: sip:x_...@192.168.4.19:2048;reg-id=1 User-Agent: snom300/7.3.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 220 v=0 o=root 641540583 641540584 IN IP4 192.168.4.19 s=call c=IN IP4 192.168.4.19 t=0 0 m=audio 52386 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv but it is never received by the server. Interestingly RINGING and REGISTER messages are working OK. The NAT router is out of our control. Are we looking at a SIP ALG getting in the way? Thanks, John -- _ -- 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
[asterisk-users] intermittent problem on 1.4
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline-voipfone-asterisk 1.4-voipfone-UK landline About 1 in 3 times the call at the final landline is silent and we see RTP Read too short scrolling on the console log. Where do we start working out what's going on? Other than that the server is working well John -- _ -- 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
[asterisk-users] vigor 2920 problems
One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- 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
Re: [asterisk-users] vigor 2920 problems
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will get permission to try new firmware later! JT On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote: Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- 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 -- _ -- 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
[asterisk-users] atx timeout - play xferfailsound
Asterisk 1.6.2.20 on Debian Lenny I'm finding that if no one answers an attended transfer (timeout set by atxfernoanswertimeout), then the transferrer is handed back to the original caller, and a beep is played. In 1.4 I was able to indicate the timeout and failure by setting xferfailsound to a custom recording, but this doesn't seem to happen in 1.6 How can I indicate a timeout to the transferrer? Many thanks John -- _ -- 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
Re: [asterisk-users] vigor 2920 problems
Thanks for help- suggestion fixed the issue John On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote: Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will get permission to try new firmware later! JT On 21 November 2011 10:45, Arthur Stanfield a...@dmcip.com wrote: Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. John -- _ -- 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 -- _ -- 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
[asterisk-users] Asterisk 1.8 not playing parking slot announcement to parker
Just upgraded to 1.8, we use the multi lot parking feature by dialling *4. We are not getting the parking slot announcement being played to the person who parks the call, so it's impossible to tell which slot they've gone into. Could someone check our config? On Debian Squeeze using packages from http://packages.asterisk.org/debsqueeze main (Asterisk 1.8.11.1-1digium1~squeeze) /etc/asterisk/features.conf [general] transferdigittimeout = 5 ; Number of seconds to wait between digits when transferring a call xfersound = beep ; to indicate an attended transfer is complete xferfailsound = unavailablebeep ; to indicate a failed transfer featuredigittimeout = 2000 ; Max time (ms) between digits for atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds. [featuremap] atxfer = *1 ; Attended transfer blindxfer = *2 ; Blind transfer (default is #) automon = *3 ; One Touch Record a.k.a. Touch Monitor parkcall = *4 ;***multitenant callparking #include /etc/asterisk/features.multiparking.conf /etc/asterisk/features.multiparking.conf [parkinglot_mhill] context = mhillpark parkpos = 1-9 findslot = first parkinghints = yes ; Add hints priorities automatically for parking slots (default is no). parkedmusicclass = classical parkingtime = 7200 parkedcalltransfers = both parkedcallreparking = both /etc/asterisk/extensions.conf ... [parkinglot_mhill] switch = Realtime/@extensions ... /etc/asterisk/sip.conf ... parkinglot=parkinglot_mhill ... -- _ -- 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
