Outgoing calls rejected b/c extension not found?

Asterisk Asterisk Support
1

Hello,
I am new to Asterisk - running it on a ASUS N66U.

[code]OS Version:
Linux RT-N66U 2.6.22.19 #4 Sun Jul 7 01:51:55 EDT 2013 mips GNU/Linux

Asterisk Build:
Asterisk/1.8.10.1
Asterisk GUI-version : SVN–rexported[/code]
The ASTERISK installation is not behind NAT.

I have incoming calls working, but get an error I don’t quite understand doing outgoing calls - 070XXXXXXX is my cell phone number;

Is it looking for a local extension instead of transfering the call to my registred SIP-trunk?

Here is the output of SIP debug on outgoing call;

[code]<— SIP read from UDP:192.168.1.164:5060 —>
INVITE sip:070XXXXXXX@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.164:5060
From: sip:6000@192.168.1.1;user=phone;tag=1469458328
To: sip:070XXXXXXX@192.168.1.1;user=phone
Call-ID: 214654148@192.168.1.164
CSeq: 1 INVITE
Contact: sip:6000@192.168.1.164:5060;user=phone;transport=udp
User-Agent: Cisco ATA 186 v3.1.1 atasip (040629A)
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Supported: replaces
Content-Length: 251
Content-Type: application/sdp

v=0
o=6000 106376 106376 IN IP4 192.168.1.164
s=ATA186 Call
c=IN IP4 192.168.1.164
t=0 0
m=audio 16384 RTP/AVP 8 0 4 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (13 headers 11 lines) —
Sending to 192.168.1.164:5060 (NAT)
Using INVITE request as basis request - 214654148@192.168.1.164
Found peer ‘6000’ for ‘6000’ from 192.168.1.164:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.164:16384
Looking for 070XXXXXXX in DLPN_Default (domain 192.168.1.1)

<— Reliably Transmitting (NAT) to 192.168.1.164:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.164:5060;received=192.168.1.164;rport=5060
From: sip:6000@192.168.1.1;user=phone;tag=1469458328
To: <sip:SIP/2.0 404 Not Found@192.168.1.1;user=phone>;tag=as23e476a4
Call-ID: 214654148@192.168.1.164
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Jul 29 20:18:07] NOTICE[5307]: chan_sip.c:22622 handle_request_invite: Call from ‘6000’ (192.168.1.164:5060) to extension ‘070XXXXXXX’ rejected because extension not found in context ‘DLPN_Default’.
Scheduling destruction of SIP dialog ‘214654148@192.168.1.164’ in 32000 ms (Method: INVITE)[/code]

Could the “SIP/2.0 404 Not Found” be my problem, or is it just a secondary symptom?

Very thankful for any help or hints to get me in the right direction.

Best regards,
Federico

2

The 404 is secondary; it is how SIP UAS’ report it to the UAC.

Your sip.conf has matched and entry which has context=DLPN_Default. That context, in extensions.conf either doesn’t contain such a context, or there is no exten => xxxxx,1,… line in that context where xxxxx is, or matches 070XXXXXXX. sip.conf includes the transitive closure of all files included from it, and any entries introduced by Asterisk realtime. Similarly for extensions.conf. The context includes any contexts included by include => lines.

3

Thank you for your reply - I could fix it by manually editing the extensions.conf.

Still not getting out though;

[code]SIP Debugging re-enabled

<— SIP read from UDP:192.168.1.226:5060 —>
INVITE sip:0708XXXXXX@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bKc742d94940dc9f1ddc4e4fbdf2b14e3f;rport
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone
Call-ID: 2234017575@192_168_1_226
CSeq: 2 INVITE
Contact: sip:6000@192.168.1.226:5060
Max-Forwards: 70
User-Agent: C610A IP/42.076.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 382

v=0
o=6000 5016 6 IN IP4 192.168.1.226
s=Mapping
c=IN IP4 192.168.1.226
t=0 0
m=audio 5016 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (14 headers 17 lines) —
Sending to 192.168.1.226:5060 (NAT)
Using INVITE request as basis request - 2234017575@192_168_1_226
Found peer ‘6000’ for ‘6000’ from 192.168.1.226:5060

<— Reliably Transmitting (NAT) to 192.168.1.226:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bKc742d94940dc9f1ddc4e4fbdf2b14e3f;received=192.168.1.226;rport=5060
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone;tag=as36b12806
Call-ID: 2234017575@192_168_1_226
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="428b9f78"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘2234017575@192_168_1_226’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.226:5060 —>
ACK sip:0708XXXXXX@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bKc742d94940dc9f1ddc4e4fbdf2b14e3f;rport
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone;tag=as36b12806
Call-ID: 2234017575@192_168_1_226
CSeq: 2 ACK
Contact: sip:6000@192.168.1.226:5060
Max-Forwards: 70
User-Agent: C610A IP/42.076.00.000.000
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.1.226:5060 —>
INVITE sip:0708XXXXXX@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bK184868e237576bc5ed0d550b5ebae6ae;rport
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone
Call-ID: 2234017575@192_168_1_226
CSeq: 3 INVITE
Contact: sip:6000@192.168.1.226:5060
Authorization: Digest username=“6000”, realm=“asterisk”, algorithm=MD5, uri="sip:0708XXXXXX@192.168.1.1;user=phone", nonce=“428b9f78”, response="6126b29bb3a96e3d5923c6034fc104c1"
Max-Forwards: 70
User-Agent: C610A IP/42.076.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 382

v=0
o=6000 5016 6 IN IP4 192.168.1.226
s=Mapping
c=IN IP4 192.168.1.226
t=0 0
m=audio 5016 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (15 headers 17 lines) —
Sending to 192.168.1.226:5060 (NAT)
Using INVITE request as basis request - 2234017575@192_168_1_226
Found peer ‘6000’ for ‘6000’ from 192.168.1.226:5060
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x191c (ulaw|alaw|g726|g729|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.226:5016
Looking for 0708XXXXXX in DLPN_Default (domain 192.168.1.1)
list_route: hop: sip:6000@192.168.1.226:5060

<— Transmitting (NAT) to 192.168.1.226:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bK184868e237576bc5ed0d550b5ebae6ae;received=192.168.1.226;rport=5060
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone
Call-ID: 2234017575@192_168_1_226
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0708XXXXXX@192.168.1.1:5060
Content-Length: 0

<------------>
Audio is at 18946
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 77.240.208.102:5060:
INVITE sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net SIP/2.0
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;branch=z9hG4bK396a2af9;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as6bb70260
To: sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net
Contact: sip:asterisk@62.63.XXX.XXX:5060
Call-ID: 4459d534466841307d8ffce3551e24b0@<my home domain - retracted>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1
Date: Sat, 17 Aug 2013 14:21:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 1467024088 1467024088 IN IP4 62.63.XXX.XXX
s=Asterisk PBX 1.8.10.1
c=IN IP4 62.63.XXX.XXX
t=0 0
m=audio 18946 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:77.240.XXX.XXX:5060 —>
SIP/2.0 100 Trying
User-Agent: Centile-Supra/1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;received=62.63.XXX.XXX;branch=z9hG4bK396a2af9;rport=5060
Content-Length: 0
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as6bb70260
To: sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 4459d534466841307d8ffce3551e24b0@<my home domain - retracted>

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:77.240.XXX.XXX:5060 —>
SIP/2.0 404 Not Found
User-Agent: Centile-Supra/1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;received=62.63.XXX.XXX;branch=z9hG4bK396a2af9;rport=5060
Content-Length: 0
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as6bb70260
To: sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 4459d534466841307d8ffce3551e24b0@<my home domain - retracted>

<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net for address/port to send to
set_destination: set destination to 77.240.208.102:5060
Transmitting (NAT) to 77.240.XXX.XXX:5060:
ACK sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net SIP/2.0
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;branch=z9hG4bK396a2af9;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as6bb70260
To: sip:0708XXXXXX@bahnhof-lda.soho1.voip.bahnhof.net
Contact: sip:asterisk@62.63.XXX.XXX:5060
Call-ID: 4459d534466841307d8ffce3551e24b0@<my home domain - retracted>
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘4459d534466841307d8ffce3551e24b0@<my home domain - retracted>’ in 32000 ms (Method: INVITE)
Audio is at 12840
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 77.240.208.102:5060:
INVITE sip:h@bahnhof-lda.soho1.voip.bahnhof.net SIP/2.0
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;branch=z9hG4bK7b36bfec;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Contact: sip:asterisk@62.63.XXX.XXX:5060
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1
Date: Sat, 17 Aug 2013 14:21:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 525207685 525207685 IN IP4 62.63.XXX.XXX
s=Asterisk PBX 1.8.10.1
c=IN IP4 62.63.XXX.XXX
t=0 0
m=audio 12840 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Scheduling destruction of SIP dialog ‘7679eea341930d752aeab60762dd97f8@<my home domain - retracted>’ in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘2234017575@192_168_1_226’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 192.168.1.226:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bK184868e237576bc5ed0d550b5ebae6ae;received=192.168.1.226;rport=5060
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone;tag=as77fe307a
Call-ID: 2234017575@192_168_1_226
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.10.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from UDP:77.240.XXX.XXX:5060 —>
SIP/2.0 100 Trying
User-Agent: Centile-Supra/1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;received=62.63.XXX.XXX;branch=z9hG4bK7b36bfec;rport=5060
Content-Length: 0
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>

<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:h@bahnhof-lda.soho1.voip.bahnhof.net for address/port to send to
set_destination: set destination to 77.240.XXX.XX:5060
Reliably Transmitting (NAT) to 77.240.XXX.XXX:5060:
CANCEL sip:h@bahnhof-lda.soho1.voip.bahnhof.net SIP/2.0
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;branch=z9hG4bK7b36bfec;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.10.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘7679eea341930d752aeab60762dd97f8@<my home domain - retracted>’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:77.240.XXX.XXX:5060 —>
SIP/2.0 404 Not Found
User-Agent: Centile-Supra/1
CSeq: 102 INVITE
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;received=62.63.XXX.XXX;branch=z9hG4bK7b36bfec;rport=5060
Content-Length: 0
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>

<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:h@bahnhof-lda.soho1.voip.bahnhof.net for address/port to send to
set_destination: set destination to 77.240.208.19:5060
Transmitting (NAT) to 77.240.XXX.XXX:5060:
ACK sip:h@bahnhof-lda.soho1.voip.bahnhof.net SIP/2.0
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;branch=z9hG4bK7b36bfec;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Contact: sip:asterisk@62.63.XXX.XXX:5060
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1
Content-Length: 0

<— SIP read from UDP:77.240.XXX.XXX:5060 —>
SIP/2.0 500 Server Internal Error
User-Agent: Centile-Supra/1
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 62.63.XXX.XXX:5060;received=62.63.XXX.XXX;branch=z9hG4bK7b36bfec;rport=5060
Content-Length: 0
From: “asterisk” <sip:asterisk@<my home domain - retracted>>;tag=as4d13fb8a
To: sip:h@bahnhof-lda.soho1.voip.bahnhof.net
Call-ID: 7679eea341930d752aeab60762dd97f8@<my home domain - retracted>

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘7679eea341930d752aeab60762dd97f8@<my home domain - retracted>’ Method: INVITE

<— SIP read from UDP:192.168.1.226:5060 —>
ACK sip:0708XXXXXX@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.226:5060;branch=z9hG4bK184868e237576bc5ed0d550b5ebae6ae;rport
From: “6000” sip:6000@192.168.1.1;tag=3578325136
To: sip:0708XXXXXX@192.168.1.1;user=phone;tag=as77fe307a
Call-ID: 2234017575@192_168_1_226
CSeq: 3 ACK
Contact: sip:6000@192.168.1.226:5060
Authorization: Digest username=“6000”, realm=“asterisk”, algorithm=MD5, uri="sip:0708XXXXXX@192.168.1.1;user=phone", nonce=“428b9f78”, response="6126b29bb3a96e3d5923c6034fc104c1"
Max-Forwards: 70
User-Agent: C610A IP/42.076.00.000.000
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘05a1f31414cb47196b2e40f304701369@<my home domain - retracted>’ Method: INVITE
Really destroying SIP dialog ‘3544865926@192_168_1_226’ Method: ACK
Really destroying SIP dialog ‘3629377735@192_168_1_226’ Method: REGISTER
[/code]

The only thing resembling an error I can find is this;

[code]SIP/2.0 401 Unauthorized

SIP/2.0 404 Not Found

SIP/2.0 503 Service Unavailable

SIP/2.0 500 Server Internal Error[/code]

I also switched to a more modern client - Siemens C610 IP - but that made no difference.

Really grateful for any feedback.

4

bahnhof-lda.soho1.voip.bahnhof.net doesn’t consider 0708XXXXXX to be a valid num ber.