Chappo

See here - http://forum.snomone.com/index.php?/topic/5846-wave-input-screaching/ Nothing out of the ordinary happening. Just selected Wave Input. Can you please provide a link to PagMOH?

As the title suggests the download for the page moh app is dead. Can we please get a link to where it can be downloaded from? Also still having issues with the WAV input screeching every 5 seconds on the latest 5.0.5. http://wiki.snomone.com/index.php?title=Miscellaneous_Downloads

You need to re-enter the license code in the License section. Had the same issue with my installation.

Alright - Will do a bit more diagnosis and see what I can come up with re the transfer issue. In regards to the IP issue it is being caused by the phones (WWW-Contact: <http://{x-snom-adr}:80>) - The first phone that was still correctly showing was on a previous release and was working - after restarting the phone yesterday and it updating it is now also not giving the correct IP address. REGISTER sip:pbx.x.com.au SIP/2.0 Via: SIP/2.0/TLS 172.28.1.82:3888;branch=z9hG4bK-40y8mt221f18;rport From: "BC" <sip:45@pbx.x.com.au>;tag=gptix9lh3z To: "BC" <sip:45@pbx.x.com.au> Call-ID: 5200000037cf-p46y1hlf9828 CSeq: 6695 REGISTER Max-Forwards: 70 Contact: <sip:45@172.28.1.82:3888;transport=tls;line=fdv44357>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:36f106c9-e8da-40de-88bd-00041345B328>" User-Agent: snom821/8.7.3.15 Allow-Events: dialog X-Real-IP: 172.28.1.82 Supported: path, gruu WWW-Contact: <http://{x-snom-adr}:80> Proxy-Require: buttons Expires: 3600 Content-Length: 0

Hi Guys, Have an issue with calls transferring when the transferring party is hanging up the handset - This is bad and has resulted in a few situations where the external calling party has been able to listen on in confidential discussions. The issue is present if the following action is taken - Calling party calls in - Answered by A. A places call on hold. Dials B via intercom *90EXT A speaks to B - B does not wish to have call transferred. A hangs up call to B and returns to external call. A places handset on hook. Call is transferred to B (and is placed onto handsfree). Using 4.5.0.1090 Epsilon Geminids (Win64) and 8.7.3.15. Transfer call on hangup is disabled. Edit: While I have this issue raised there also appears to be an issue with the display of client IP addresses within the SnomOne console. The first extension IP is correct however all subsequent IP's are listed as "http://%7Bx-snom-adr%7D/".

We have already done that. That resolves the issue of the phone CallerID being updated but there is not an update issued via http (Action URL) to the software we have running on the PC.

Guys, How is the new version coming along? Copping some flack here due to this bug being outstanding still.

They were removed so to prevent the forum code from messing up the http. That's good to know. Any idea of time frames?

Any news?

That http post is sent from the phone to the client. This would appear to be an issue with the phone firmware and not the SnomOne. You shouldn't need to have the software installed. The below settings should give you the same result as it should still send the post result. action_dnd_on_url!: 172.28.1.81:4052/action_dnd_on_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote action_dnd_off_url!: 172.28.1.81:4052/action_dnd_off_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote action_incoming_url!: 172.28.1.81:4052/action_incoming_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote action_outgoing_url!: 172.28.1.81:4052/action_outgoing_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote action_connected_url!: 172.28.1.81:4052/action_connected_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote action_disconnected_url!: 172.28.1.81:4052/action_disconnected_url?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote

Is there any more information that is needed to get this resolved?

The problem from what I can gather is that the http action_connected_url is getting sent before the info message is being received - If this case no matter what software is in use on the PC it will not have the correct CLID. You can see in the logs that also each time that the phone call is placed on hold and retrieved that the CLID is the pickup star code again. Only when going into the hold or disconnected state and firing off the action_disconnected_url (and action_hold_url) that the CLID was correct - Which by this state is too late to update the CLID. Unfortunately no I do not have a log from when it was previously working.

The updated version still has not resolved the issue with Contact Pad / Flexor. I have been doing some more logging and have determined what the issue appears to be - This is more evident when answering an external call (It is just not occurring on attended transfers). As you can see from the logfile and SIP trace the initial http post sent to the PC includes a disprem with the star code pickup. It is only after a few seconds that the INFO message is received with the correct CLID. Sending post request host = 172.28.1.77:4052, file = /action_connected_url?local=45%40pbx.domain.com.au&remote=%2a6017&callid=08a9273c6eda-9rycw0bd21p6&csta=8&active_url=45%40pbx.domain.com.au&active_user=45&active_host=pbx.sgk.com.au&displocal=BC&disprem=%2a6017 The $display_remote key appears to be initially using the incorrect value. The disconnect action url has the correct CLID however by that stage it is too late. Edit: I have also attached the SnomONE log which correctly shows the CLID on the inbound call. Ex45 SIP Trace.txt Ext45 LOG.txt log-2012-06-21 - SnomONE Log.txt

Thanks Katerina. Yes we would be willing to test. Using Server 2008 R2.

I have attached a SIP trace of an attended transfer. The issue is as follows:- We are currently using Flexor Manager (But have also had the same issue with ContactPad) in which when an attended transfer is sent through to an extension the software only reports the internal caller which transferred the call through - It does not update for the external parties number. The attended_transfer action item is shown as - 172.28.1.73:4052/action_attended_transfer?local=$local&remote=$remote&callid=$call-id&csta=$csta_id&active_url=$active_url&active_user=$active_user&active_host=$active_host&displocal=$display_local&disprem=$display_remote We had the same issue on the phones where the external number would not show but changing the ignore_asserted_in_gui allows for the correct update - I presume that the "asserted number" is what is being passed to the Flexor/ContactPad software. It has only been an issue with the last couple of snomone releases. Is this what internal fault ID SONE-186 has been lodged for? SnomLog.txt

Thanks Katerina, Can you please advise when you expect a release that fixes this issue? The CTI software that we are using is crippled whilst this issue is occurring.

Thanks Guys - I am using the PBX behind NAT with the only IP address allowed through on 5060 is the IP of the SIP provider. That said we have worked out what was causing one of the companies calls to fail - They had a malformed CLID which was two digits short and my provider was rejecting the calls.

I had increased the number of ports to 2000 however I was still getting errors. Having a look at the open ports on the server I found that the dns system had about 2500 ports reserved for it. And the range that I selected for the RTP ports was smack bang in the middle. Port errors are now resolved however I am still curious as the origins of 65.19.131.47? It appears to be a Hurricane Electric IP in the US? Call port 130: update_codecs for af981122845@65.19.131.47: codec_preference size 4, available codecs list is empty

Thanks - There is a total of 4 available codecs within that section. I've attached a screenshot. I am also curious to know why I am getting the unable to allocate port error message too? Edit: Also where is it getting this IP from? 65.19.131.47? It has nothing to do with my SIP provider?

Hey Guys, I am having an intermittent issue with inbound calls. There is a few companies that are unable to call through. When this happens I am receiving a call to my mobile phone - These calls are from a Private number (All outbound trunks have CLID enabled). When I answer the call there is no voice/sound. I believe this to the failover that I have configured with my VOIP provider when the 4 VOIP lines are unreachable. 2012-05-10T16:25:29 0:00:07 Australia - Mobile Fixed to Mobile 9406156 XXXXXXXX -0.15 Outbound 2012-05-09T13:34:15 0:00:18 Australia - Mobile Fixed to Mobile 9406156 XXXXXXXX -0.15 Outbound 2012-05-07T14:14:06 0:00:04 Australia - Mobile Fixed to Mobile 9406156 XXXXXXXX -0.15 Outbound 2012-05-02T16:44:33 0:00:05 Australia - Mobile Fixed to Mobile 9406156 XXXXXXXX -0.15 Outbound 2012-05-02T12:45:36 0:00:03 Australia - Mobile Fixed to Mobile 9406156 XXXXXXXX -0.15 Outbound When checking the logs at these times I have found a common occurrence - The issues seems to be from a lack of available codecs - For what reason this is occurring I am not quite sure? Also I have configured my Snom Installation so that the only SIP server that can connect is as follows: AssociatedAddresses: sip00.mynetfone.com.au. Why am I getting the calls that appear to be coming through from 65.19.131.47? 20120510142529: Call port 130: update_codecs for af981122845@65.19.131.47: codec_preference size 4, available codecs list is empty [5] 20120510142529: Available codec list is empty for af981122845@65.19.131.47 [5] 20120510142529: SIP Tx udp:125.213.160.81:5060: [1] 20120510142529: Could not allocate new ports! [5] 20120510142529: SIP Tx tls:172.28.1.68:3150: [5] 20120510142529: SIP Rx udp:125.213.160.81:5060: [3] 20120509113416: Call port 12: update_codecs for ae1028169497@65.19.131.47: codec_preference size 4, available codecs list is empty [5] 20120509113416: Available codec list is empty for ae1028169497@65.19.131.47 [5] 20120509113416: SIP Tx udp:125.213.160.81:5060: [1] 20120509113416: Could not allocate new ports! [5] 20120509113416: SIP Tx tls:172.28.1.68:3855: [5] 20120509113416: SIP Rx udp:125.213.160.81:5060: [5] 20120509113416: SIP Rx udp:125.213.160.81:5060: [1] 20120509113416: Could not allocate new ports! 20120502104536: Call port 218: update_codecs for ae125644798@65.19.131.47: codec_preference size 4, available codecs list is empty [5] 20120502104536: Available codec list is empty for ae125644798@65.19.131.47 [5] 20120502104536: SIP Tx udp:125.213.160.81:5060: [1] 20120502104536: Could not allocate new ports! [5] 20120502104536: SIP Tx tls:172.28.1.68:3887: [5] 20120502104536: SIP Rx udp:125.213.160.81:5060: [5] 20120502104536: SIP Tx tls:172.28.1.68:3887: [3] 20120502144433: Call port 345: update_codecs for ac168578346@65.19.131.47: codec_preference size 4, available codecs list is empty [5] 20120502144433: Available codec list is empty for ac168578346@65.19.131.47 [5] 20120502144433: SIP Tx udp:125.213.160.81:5060: [1] 20120502144433: Could not allocate new ports! [5] 20120502144433: SIP Tx tls:172.28.1.68:3887: [5] 20120502144433: SIP Rx udp:125.213.160.81:5060: [5] 20120502144433: SIP Tx tls:172.28.1.68:3887:

Thank you Kat. That has allowed the phones to correctly display the Caller ID - However using ContactPad and/or Flexor Manager the Caller ID is not being updated. It is still showing as the extension that transferred the call. Is there anything else that can be done short of a snomone update to fix the issue?

There is no INFO messages in either the server log or the phone SIP log.

Guys, As per the title I am having issues with the Caller ID updating when an attended transfer occurs. When looking at the SIP trace on the receiving handset the P-Asserted Identity is always the person that initiated the transfer. It is not updating to be the external trunk ID. Received from tls:172.28.1.5:5061 at 22/3/2012 13:21:06:670 (489 bytes): PRACK sip:45@172.28.1.82:3120;transport=tls;line=8gg2i362 SIP/2.0 Via: SIP/2.0/TLS 172.28.1.5:5061;branch=z9hG4bK-786e1c01f663a1a0dcfcacc96a406222;rport From: "AS" <sip:40@pbx.domain.com.au>;tag=60344 To: "BC" <sip:45@pbx.domain.com.au>;tag=1b32s1jnpo Call-ID: f52b415a@pbx CSeq: 31620 PRACK Max-Forwards: 70 Contact: <sip:45@172.28.1.5:5061;transport=tls> RAck: 1 31619 INVITE P-Asserted-Identity: "AS" <sip:40@pbx.domain.com.au> Content-Length: 0 Current version info (Have also tried the latest beta with no change) Version: 2011-4.5.0.1016 Alpha Monocerotids (Win64) Created on: Dec 22 2011 14:15:39 License Status: snom ONE yellow