Hello good morning,
On this moment I have this configuration:
Asterisk <----> Cisco Call Manager <–MGCP–> Gateway PRI Cisco 2801
|
|<---------MGCP----------> Gateway FXS Cisco VG224
On my Asterisk PBX I have 650 IP-phones registered and I want to migrate the VG224 to Asterisk.
And when I migrate the VG224 I want to migrate 2801 and swicth off the CCM
I have had several problems with VG224:
- When I tried to register 1 X FXS port of my VG224 on my Asterisk the port didnt register with this config:
sip.conf:
[77001]
type = friend
host =dynamic
secret=pepito
dtmfmode=rfc2833
canreinvite=no
context=from-internal
trustrpid=yes
sendrpid=no
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
callcounter=yes
faxdetect=no
If I do the same with a Grandstream 4224 (for example) I can register the extension without any problems.
In the VG224 I created a dial peer associated to extension 77001 with pass pepito and if I executed CLI> sip show peers ,the exension appears with UNKNOWN status
If I change host=dynamic to host = IP of VG224 then the status change to OK
Any idea about I needed to change the host from dynamic to IP of VG224 to can register the VG224 port on my Asterisk IPPBX?
- When I tried to do calls from an IP phone to an analog phone which was connected on VG224 (both phones registered on my Asterisk-IPPBX) it appeared an error which said: 400 Bad Request ‘Malformed/Missing…’
I did can observe that when I called to 77001 ext., on the INVITE message the field To was created on a bad manner:
To: sip:DireccionIPCentralita;transport=UDP
instead of
To: sip:[email protected];transport=UDP
If I add username = 77001 on sip.conf this problem disaapeared.
¿Any idea about why I need to use the username field? With grandstream 4224 I didnt have that kind of poblems
When I do calls between VOIP Phones I dont need to add the username parameter and the INVITE send the To:<… field good
- And the last problem is related with faxes:
If I connect an analog phone to the FXS port I can make internal/outbound calls and receive incoming calls without problems.
If I connect a FAX on the same FXS port:
I can call to the FAX from a phone and the FAX offhook.
I can call to the FAX from another FAX, then the FAX offhook and finally gives an error message.
If I try to call from the FAX the call is not cursed, even doesnt appears on the asterisk console ¿It does any sense? I think that at least it should appear a triying of call on the CLI (although the call finally can give an error)
Maybe the VG224 config that Im using is not correct. On vg224 I have created a dial peer with:
- modem passthrough nse codec g711alaw
- vad deactivated
- echo canc. deactivate
- dtmf-relay voip codec all mode nte-ca
I have not limited the fax speed.
Best regards and thanks by your help
Miguel Sanz
