amygrant
Guru
- Joined
- Apr 4, 2010
- Messages
- 132
- Reaction score
- 3
Had a perfectly working PIAF purple 1.7.5.5 server running Asterisk 1.8.0. Had multiple gvoice accounts, skype as a trunk, and traveling man working all working great.
Had to move this server to a different location with a different IP behind a different router with a different private addressing namespace.
Incoming calls via skype DID or GV DID or sipgate DID all work as far as sending the caller to the correct IVR or voice mail. SIP trunks register properly. External SIP phones register properly.
My problem occurs when I try using an external SIP device to make any sort of call to another external SIP device or to make a call to a phone number using either skype as a trunk or GV.
On my external soft phone (xlite on a desktop) I try to call my cell phone and use my GV trunk. My cell phone rings, I get 2 way audio but after a few seconds the call is dropped. In my log I see this:
If I don't answer my cell quickly, the call will still be dropped with a similar timeout error.
If I try using my external desktop xlite to call my laptop xlite, which is also outside the asterisk network, the laptop never rings and I get a similar timeout error on the asterisk server. If I try using my laptop xlite to call my desktop xlite the same error occurs. If I try using either xlite to dial an IVR or voicemail extension, however, it works. So anything internal to the askterisk server seems to work but anything heading out the asterisk network fails.
So what changed?
1. The new location of the asterisk server has a new router (Netgear WNR1000v2) whereas before I had a FreeBSD server acting as a gateway.
This is a brand new router that was purchased. Simple setup cable modem -> netgear -> asterisk server
The netgear has the MAC address of the asterisk server so the netgear's DHCP always assigns the asterisk server the private IP address of 192.168.1.5
The netgear has port forwarding setup of 5060, 10000-20000, and 5222 all to 192.168.1.5. It has some other ports open for SSH and SSL.
2. The private namespace has changed from 10.0.0.x to 192.168.1.x
In sip_custom.conf, I put
3. The old router had a static IP whereas the new one is dynamic. As a result, I am using dyndns.
In sip_custom.conf, I put
I am running ddcleint on the asterisk server. I am NOT using the dynamic DNS feature of the netgear router. On the asterisk server, if I ping my dyndns subdomain I get the correct outside IP.
The configuration for my external SIP extensions includes
I do not have a sip device internal to the asterisk network and unfortunately it is now in another state so I am trying to resolve this issue remotely.
After researching on google and reading the document the error message recommended, this is a network issue and I can only assume it has to do with the new network topology, or the router, or the new use of dyndns.
I am posting this here for suggestions but also for future reference in case someone else has the same problem.
I have checked everything I can think of to make sure all files have been updated to reflect the new network topology. The fact that calls from external SIP to cellphone sometimes make it, albeit only for a few seconds, has me thinking this may be cause by some sort of quality control mechanism on the asterisk end since if the networking was screwed up, I would think that there would be consistent failure as opposed to sometimes working briefly.
Any recommendations? Should I have my router be handling the dyndns and turn off ddclient on asterisk?
Had to move this server to a different location with a different IP behind a different router with a different private addressing namespace.
Incoming calls via skype DID or GV DID or sipgate DID all work as far as sending the caller to the correct IVR or voice mail. SIP trunks register properly. External SIP phones register properly.
My problem occurs when I try using an external SIP device to make any sort of call to another external SIP device or to make a call to a phone number using either skype as a trunk or GV.
On my external soft phone (xlite on a desktop) I try to call my cell phone and use my GV trunk. My cell phone rings, I get 2 way audio but after a few seconds the call is dropped. In my log I see this:
Code:
[2011-01-01 09:46:42] WARNING[25890] chan_sip.c: Retransmission timeout reached on transmission MmQyY2I1MjdmODEwODMwYjdjNWFhNGUyNjA3YzM0N2Y. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
Packet timed out after 19392ms with no response
[2011-01-01 09:46:42] WARNING[25890] chan_sip.c: Hanging up call MmQyY2I1MjdmODEwODMwYjdjNWFhNGUyNjA3YzM0N2Y. - no reply to our critical packet (see doc/sip-retransmit.txt).
If I don't answer my cell quickly, the call will still be dropped with a similar timeout error.
If I try using my external desktop xlite to call my laptop xlite, which is also outside the asterisk network, the laptop never rings and I get a similar timeout error on the asterisk server. If I try using my laptop xlite to call my desktop xlite the same error occurs. If I try using either xlite to dial an IVR or voicemail extension, however, it works. So anything internal to the askterisk server seems to work but anything heading out the asterisk network fails.
So what changed?
1. The new location of the asterisk server has a new router (Netgear WNR1000v2) whereas before I had a FreeBSD server acting as a gateway.
This is a brand new router that was purchased. Simple setup cable modem -> netgear -> asterisk server
The netgear has the MAC address of the asterisk server so the netgear's DHCP always assigns the asterisk server the private IP address of 192.168.1.5
The netgear has port forwarding setup of 5060, 10000-20000, and 5222 all to 192.168.1.5. It has some other ports open for SSH and SSL.
2. The private namespace has changed from 10.0.0.x to 192.168.1.x
In sip_custom.conf, I put
Code:
localnet=192.168.1.0/255.255.255.0
3. The old router had a static IP whereas the new one is dynamic. As a result, I am using dyndns.
In sip_custom.conf, I put
Code:
externhost=something.dyndns.biz
I am running ddcleint on the asterisk server. I am NOT using the dynamic DNS feature of the netgear router. On the asterisk server, if I ping my dyndns subdomain I get the correct outside IP.
The configuration for my external SIP extensions includes
Code:
host=dynamic
canreinvite=no
nat=yes
I do not have a sip device internal to the asterisk network and unfortunately it is now in another state so I am trying to resolve this issue remotely.
After researching on google and reading the document the error message recommended, this is a network issue and I can only assume it has to do with the new network topology, or the router, or the new use of dyndns.
I am posting this here for suggestions but also for future reference in case someone else has the same problem.
I have checked everything I can think of to make sure all files have been updated to reflect the new network topology. The fact that calls from external SIP to cellphone sometimes make it, albeit only for a few seconds, has me thinking this may be cause by some sort of quality control mechanism on the asterisk end since if the networking was screwed up, I would think that there would be consistent failure as opposed to sometimes working briefly.
Any recommendations? Should I have my router be handling the dyndns and turn off ddclient on asterisk?