shockingblue
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Posts posted by shockingblue
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Hi. Tested, again, to call my home number from a cell phone. Trunk sends call to hunt group (!(\+46731234567)!\1!f!70! !([a-z]{2,9})!\1!t! !^(4[0-9]{1})!\1!t! 72) and the phones start ringing. I hang up before the call goes to voicemail. There is no trace of the missed call in the users portal, but on all Snom 320 phones (in the hunt group) the missed call is listed. I would like "only" the missed calls from a Snom 320, in the simplest way, on a seperate webpage or tab. Preferably, but not nesseceraly, in the pbx user portal for easy reading, but anyway goes. How?
The only way missed calls shows in the user portal of the pbx under Missed Calls is if I call an extension directy (e.g. 45).
I'm using 2011-4.5.0.1016 Alpha Monocerotids (Win32) cause I like it and it's the latest free version I downloaded.
I would like to get hold of later 4.5.x builds but they are no longer downloadable.
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Hi
Since missed calls don't show in the pbx user login web interface (Call History/ Missed Calls) for phones in a hunt group I'd like to show the phones (snom 320) own missed calls record in the pbx user login web interface. I've created a new submenu next to Call Log and Missed Calls under the tab Call History in the pbx user login web interface by modifying the usr_ templates and would like to populate it with missed calls data from the phone. How is this doable with or without getting dirty with php and/ or installing a web server? I'm using 2011-4.5.0.1016 Alpha Monocerotids (Win32). Thanks...
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I'm very impressed with Snom One after 2 weeks of newbie testing. Maybe I'm using it in a way not intended i.e. for phone numbers and SIP URI's alike.
Unfortunately there's an inbound problem. Trunk replaces the callers domain name with my domain name when someone calls my SIP address!?
So, if bill@microsoft.com call myname@mydomain.dyndns.org then my Snom phone will show Missed Calls from bill@mydomain.dyndns.org.
Please look at the From: "Bill Gates" headers below, in the Snom One Logfile, how the domain name changes. Most grateful if someone can remedy that.
[7] 2011/10/28 09:08:21: SIP Rx udp:80.190.40.50:5060:
INVITE sip:myname@mydomain.dyndns.org SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK77e68e782c52808d24a5a5c3d2f3b6b0.0
Via: SIP/2.0/UDP 192.168.50.31:42389;branch=z9hG4bKPjyWhrtEnPZ6IG0HUIqZ8D1J8PzgHTbdru;received=192.168.50.31;rport=42389
Max-Forwards: 69
From: "Bill Gates" <sip:bill@microsoft.com>;tag=34TnZZHGX6oQ7AsqzyNWGULE4YA.On6k
To: sip:myname@mydomain.dyndns.org
Contact: "Bill Gates" <sip:bill@192.168.50.31:42682>
Call-ID: CIcZ.2iuSBYi3HrQGkfqKRH6eS2UNpCU
CSeq: 14713 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria Android 1.1.8
Content-Type: application/sdp
Content-Length: 253
Record-Route: <sip:25c260081bdf382f@192.168.0.1;lr=true>
v=0
o=- 3528781776 3528781776 IN IP4 192.168.50.31
s=cpc_med
c=IN IP4 192.168.50.31
t=0 0
m=audio 4000 RTP/AVP 18 0 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
[5] 2011/10/28 09:08:21: Identify trunk (IP address and DID match) 2
[7] 2011/10/28 09:08:21: SIP Tx udp:80.190.40.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK77e68e782c52808d24a5a5c3d2f3b6b0.0;rport=5060;received=80.190.40.50
Via: SIP/2.0/UDP 192.168.50.31:42389;branch=z9hG4bKPjyWhrtEnPZ6IG0HUIqZ8D1J8PzgHTbdru;received=192.168.50.31;rport=42389
Record-Route: <sip:25c260081bdf382f@192.168.0.1;lr=true>
From: "Bill Gates" <sip:bill@microsoft.com>;tag=34TnZZHGX6oQ7AsqzyNWGULE4YA.On6k
To: <sip:myname@mydomain.dyndns.org>;tag=68a86c6fa8
Call-ID: CIcZ.2iuSBYi3HrQGkfqKRH6eS2UNpCU
CSeq: 14713 INVITE
Content-Length: 0
[6] 2011/10/28 09:08:21: Sending RTP for CIcZ.2iuSBYi3HrQGkfqKRH6eS2UNpCU to 192.168.50.31:4000, codec not set yet
[5] 2011/10/28 09:08:21: Domain trunk Netatonce@mydomain.dyndns.org sends call to 41 in domain mydomain.dyndns.org
[7] 2011/10/28 09:08:21: set_codecs: for CIcZ.2iuSBYi3HrQGkfqKRH6eS2UNpCU codecs "", codec_preference count 6
[7] 2011/10/28 09:08:21: set_codecs: for 426603d7@pbx codecs "", codec_preference count 6
[7] 2011/10/28 09:08:21: SIP Tx udp:192.168.0.31:2098:
INVITE sip:41@192.168.0.31:2098;line=bcxudu4l SIP/2.0
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK-abecbe0e54d289877a733a0385aecd02;rport
From: "Bill Gates" <sip:bill@mydomain.dyndns.org;user=phone>;tag=27972
To: "My Name" <sip:41@mydomain.dyndns.org>
Call-ID: 426603d7@pbx
CSeq: 3786 INVITE
Max-Forwards: 70
Contact: <sip:41@192.168.0.33:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.1.4025
Alert-Info: <http://127.0.0.1/Bellcore-dr3>
Content-Type: application/sdp
Content-Length: 325
v=0
o=- 5619 5619 IN IP4 192.168.0.33
s=-
c=IN IP4 192.168.0.33
t=0 0
m=audio 57346 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[7] 2011/10/28 09:08:21: SIP Rx udp:192.168.0.31:2098:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.33:5060;branch=z9hG4bK-abecbe0e54d289877a733a0385aecd02;rport=5060
From: "Bill Gates" <sip:bill@mydomain.dyndns.org;user=phone>;tag=27972
To: "My Name" <sip:41@mydomain.dyndns.org>;tag=olrnw5vfg5
Call-ID: 426603d7@pbx
CSeq: 3786 INVITE
Contact: <sip:41@192.168.0.31:2098;line=bcxudu4l>;reg-id=1
Require: 100rel
RSeq: 1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0
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Did you also assign the dial plan to the domain and/or to the extension?
Yes, I've tried both. Can't figure out how to build a dial plan that doesn't modify the SIP URI (info@intertex.se) dialed. It works fine for calling ordinary phone numbers with the simplest dial plan (pattern: *, replacement: empty). Calling i.e. 90510 gets converted to sip:90510@sip1.netatonce.net;user=phone which is fine.
However, a complete SIP URI i.e. 9902@qxip.net doesn't work cause the dial plan changes it to sip:9902@sip1.netatonce.net;user=phone (argh!). It replaces the domain (qxip.net) with the domain name of the SIP Registration Trunk (Generic SIP Server) of my broadband supplier and ITSP, Netatonce. Calling info@intertex.se doesn't work at all (SIP/2.0 404 Not Found) with the simple dial plan as above. Note: sip:info@intertex.se is not in use anymore.
I've tried |^([a-zA-Z0-9&=+\$,;?\-_.!~*‘()%]+@.+)|sip:\1| which gives nothing, unless I remove the "!" in the string. Then I get sip:info@intertex.se@sip1.netatonce.net;user=phone when all I want is info@intertex.se. If I register a Snom 320 directly on my ITSP's SIP server (sip1.netatonce.net) both phone numbers (routed to PSTN) and SIP URI's i.e. info@intertex.se works. Confused...
I would appreciate some, down to earth, guidance on how to build a simple dial plan that adds "@domain;user=phone" when a phone number is dialed but doesn't modify a complete SIP URI (an address containing the_@_sign). My country code is 46 (Sweden) and area code 40 (Malmoe). Thanks in advance... have a great day!
Update: After some testing my current, SIP URI capable, dial plan looks like this. Cool, because now recording works when calling a SIP URI
10;Netatonce;;112;
20;Netatonce;;^([0-9]{5,});
30;Netatonce;;"^([a-zA-Z0-9&=+\$,;?\-_.~*‘()%]+@.+)";sip:\1 (sip:\1 is the replacement)
90;Netatonce;;*;
Not bulletproof, but it works if the SIP URI doesn't start with more than four digits. Hoping that someone can improve pos. 20 in the dial plan or better the whole piece while you're at it.
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Hi all!
pbx newbie here. I can't find out how to get a SIP URI through Snom One without getting changed.
Dial plan seems to have been built without the accomodation for SIP dialing to be done through it?
Please see http://macnugget.org/projects/asterisk/page7
My current dial plan (to keep thing simple) is: Pattern: * , Replacement: empty
Guess I'm after the same thing as shopcomputer?
Missed Calls
in Web Interface
Posted
Copy that, but how about extracting (web get, action url or something) the missed calls from a Snom 320 to a file on the drive. Preferably to a file in the html directory of the pbx. From there I can load it into a custom made page or tab in the pbx user portal. Doesn't someone know how to get, extract or make the phone post the missed calls record when it gets updated?