Hi,
I have some Cisco 7911 phones which I used succesfully before with a basic Asterisk setup (Arch Linux + Asterisk). I’m trying out the FreePBX distro now because I have a Sangoma card which I want to work out of the box without driver hassle.
The phones register fine via SIP. I used the extensions app in FreePBX to make two extensions (201 and 202). However when I try to make an internal call between the phones, it seems Asterisk is trying call immediately after punching in the first digit (any digit, and even hashtag and pound) and not waiting for other input.
Log:
<--- Reliably Transmitting (no NAT) to 10.30.6.83:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.30.6.83:5060;branch=z9hG4bKe9eba770;received=10.30.6.83
From: "201" <sip:[email protected]>;tag=00215552e95b00391290dd92-9fc7a399
To: <sip:[email protected];user=phone>;tag=as793f2795
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-12.0.21(11.14.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6438b016"
Content-Length: 0
<------------>
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Sending to 10.30.6.83:5060 (no NAT)
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] netsock2.c: == Using SIP RTP TOS bits 184
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] netsock2.c: == Using SIP RTP CoS mark 5
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 0
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 8
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 18
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 102
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 116
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found RTP audio format 101
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format PCMA for ID 8
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format G729 for ID 18
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format L16 for ID 102
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format iLBC for ID 116
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g729|ilbc|slin16)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Peer audio RTP is at port 10.30.6.83:19544
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c: Looking for 2 in from-internal (domain 10.30.0.113)
[2015-01-06 10:05:55] VERBOSE[3045][C-0000000c] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 10.30.6.83:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.30.6.83:5060;branch=z9hG4bK79450525;received=10.30.6.83
From: "201" <sip:[email protected]>;tag=00215552e95b00391290dd92-9fc7a399
To: <sip:[email protected];user=phone>;tag=as793f2795
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-12.0.21(11.14.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
As you can see, it’s looking for extension “2” and not waiting for me to punch in the rest. Can anyone tell me how to go about solving this? Is it a problem with the dialplan? I would expet the verbose output in the console to show the dialplan execution if it actually matches something. I didn’t have this issue with a standard Asterisk install and my experience is limited. I’m just trying to provide a simple working setup for a small non-profit.
Thanks for any assistance.