sudo
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Posts posted by sudo
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sounds like the call never went out... are you able to call the cell phone by dialing it directly? Can you copy/paste the log file of the failing scenario on this thread
http://wiki.snomone.com/index.php?title=Snom_ONE_log
Thanks
I tried sending you a personal message and I got this error:
The member MR X cannot receive any new messages
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When I dial *00(Extensoin #) there is an immediate response form the pbx that says, "There is no answer to your request. To receive a call back..."
I did migrate this domain off an old pbxnsip box. Perhaps there is an old setting that carried over?
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Version: 5.0.10 (CentOS64)
I assume you mean try calling the cell phone with the starcode?
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Ive got a problem on one of my domains.
The system will not forward out either by cell phone rules or 'forward all calls to'.
Testing on other domains it works fine, but something on this domain is broken.
My cell settings are as follows:
Include the cell phone in calls to extension:Immediately
When calling the extension in a hunt group:Include cell phone in the group
When calling the extension in an agent group:Include cell phone in the group
As I said before, This is failing on both the cell phone redirect and the "Call forward all calls to:"
What could be breaking this on this one domain?
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Any thoughts on getting the CID to show the username instead or along side with the ANI?
Ive done some more fiddeling with the CAller ID settings and have not got anything to work.
Thanks in advance.
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I modified "SIP Caller-ID Presentation" under the trunk settings to "RFC3325 P-asserted-identification" and im now getting the DID where before it said "No Caller ID"
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This must be someting on the pbx. I have been migrating customers off an old pbxnsip box. The old box was displaying the correct CID but now that I have migrated it to this ver. 5 box im having issues. Ill look over the trunk settings and see what happens.
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AFAIK CNAM (which is not CNAME) can not be signaled on per call bases and must be set up somewhere deep in the carriers network. At least we are currently struggling that out number shows "Vodia Networks" (for weeks already).
You can easily control the "display name" of the trunk in the trunk setting. Maybe just try out and see if it makes any difference before you spend too much time on something that you cannot control anyway through the trunk.
Word
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Since upgrading to ver 5 I have a couple of customers that noticed that their caller id is reading "Unknown"
I have checked with my upstream provider and all the appropriate CNAM entries have been completed but the caller id is not displaying that info
Is there somewhere on the pbx that I can set this up? Perhaps theres a setting blocking the CID info?
Anyone got any ideas?
Sudo
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Thanks JTEL,
I appended the below stanza to the end of polycom_all.xml:
voice.volume.persist.handset="1"voice.volume.persist.headset="1"voice.volume.persist.handsfree="1"Works now. -
You will have to add this by yourself... The new snom ONE Polycom templates contains only the things that should be different from the defaults.
Well the defaults in the .cfg file is set to have this persist
. So adding the stanza should result in the same .cfg as it is now. And it currently not perisiting. -
We had that in the old times (pbxnsip) as a provisioning parameter. Polycom has the default to restore the volume--I guess that makes sense because the volume differs significantly depending on who is calling (device) and who is talking. Also it helps to keep the system in a kind of calibrated state. But of course you can change the polycom provisioning template and change the setting.
Im not seeing the parameter "<volume voice.volume.persist.handset="1..." in the polycom_master.xml or polycom_all.xml
Am in in the wrong file? Is it a different string than the one above?
Thanks
Sudo
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Im having a similar issue. My Polcom IP331 will not keep the set incall valume. It keeps going back to the default setting after every call.
The polycom_sip.cfg file has the following parameter set:
<volume voice.volume.persist.handset="1" voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/>
This was the default. In all my research on Polycom forums, it states that this enables the volume to persist.
However my incall volume is not persisting. As stated before it goes back to the defualt value after every call.
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Duh....I dont know how I missed that
lol.Thanks!
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Since the upgrade to ver 5.0.10 I cant find where to make an extension the domain admin. I can create a system administrator in the global Security settings, but I can t find where in the domain to make an extension a domain admin.
Easy question, I know, but I have not been able to find the answer online.
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We had reports that changing from TLS to TCP dramatically increased the stability of installations. We are also scratching our heads why. It is easy to blame US NSA and the British Intelligence for this, but no serious explanation. We need to keep watching this topic.
Haha - Can that be our 'Get out of jail free" card? Just tell the customer that big brother is messing up their encrypted sip traffic. 
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Anyone? I dont know why this is so inconsistent?
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I appended the above string to the beginning of the stanza and it worked. Thanks for your help!
For reference the working stanza is as follows:
{loop-start}
reg.{lc}.lineKeys="{lines}"
reg.{lc}.address="{account}"
reg.{lc}.auth.password="{password}"
reg.{lc}.auth.userId="{account}"
reg.{lc}.label="{account}"
reg.{lc}.auth.domain="{domain}"
reg.{lc}.displayName="{display-name}"
reg.{lc}.outboundProxy.address="{ip-adr}"
reg.{lc}.outboundProxy.port="{sip-port tcp}"
reg.{lc}.outboundProxy.transport="TCPOnly"
reg.{lc}.serverFeatureControl.signalingMethod="subscribeAsFeatureEvent"
reg.{lc}.serverFeatureControl.subscribeToUri="{account}"
reg.{lc}.server.1.address="{domain}"
reg.{lc}.server.1.port="{sip-port tcp}"
reg.{lc}.server.1.register="1"
reg.{lc}.server.1.transport="TCPOnly"
{loop-end}
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I believe what is missing is this: reg.{lc}.lineKeys="{lines}"
Can you put that into context? Where in the above stanza should that be added? Like i said, im not a programmer.
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I see in the polycom_all.xml:
{loop-start}
reg.{lc}.address="{account}"
reg.{lc}.auth.password="{password}"
reg.{lc}.auth.userId="{account}"
reg.{lc}.label="{account}"
reg.{lc}.auth.domain="{domain}"
reg.{lc}.displayName="{display-name}"
reg.{lc}.outboundProxy.address="{ip-adr}"
reg.{lc}.outboundProxy.port="{sip-port tcp}"
reg.{lc}.outboundProxy.transport="TCPOnly"
reg.{lc}.serverFeatureControl.signalingMethod="subscribeAsFeatureEvent"
reg.{lc}.serverFeatureControl.subscribeToUri="{account}"
reg.{lc}.server.1.address="{domain}"
reg.{lc}.server.1.port="{sip-port tcp}"
reg.{lc}.server.1.register="1"
reg.{lc}.server.1.transport="TCPOnly"
{loop-end}
Im no programmer but I believe there is an issue with "{lc}" (line count?) seems like this may be the broken piece.
I assume the this script pulls the line count from the snomone gui config. Do you think im close? How do we fix this?
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I just manually entered the line into in the WebUI of the polycom and it came up with no issues.
This leads me to think that my Polycom PnP .xml might be broken.
Anyone know what .xml controls the line configuration?
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Ohhhhh.... Okay now I understand you. I thought you wanted difference extensions on the same phone.
So what you want are several private lines on the phone. You actually have that by default, even though there is only one button associated with it. Did you try setting the "lines" parameters in the registrations tab? At least in the old pbxnsip days, that provisioned several lines on the phone; though with the latest version that could conflict with the explicit button assignments (but worth a quick try). But again, I would personally not put several lines on the precious buttons on the phone, instead use the phone user interface to toggle between the different calls on the phone.
I thought that I was clear, but apparently not. My apologies.
You have hit the nail on the head as that is exactly my problem. I add more lines in the "lines" field under the registrations tab and it still only shows up with one line on the phone. The phone is configured via PnP. This was working great before and now its broken.
The webui is a great option, but this particular cust is specifically asking for them back on the phone.
I appreciate your advice. Do you have any more suggestions?
~Sudo
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Well you need to remove and add the MAC to all extensions that share the same MAC. But apart from that, you did it right. The generated folder is not read, only written and should help you figure out what was sent to the device.
You can double check the macs directory if the PBX got it right.
Checking the macs/ dir, there is only 1 corresponding .xml. If I understand correctly, I should see more than one to create this conflict (?). This MAC is not used by any other extension. I do see that this MAC is in 2 different extensions in the generated/ dir but this should not matter as the generated dir is only written to.
This phone is only going to be associated with one extension and I need multiple lines on it.
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The problem is related to the "last one who saves the MAC wins" problem that we have already discovered and even fixed. It is already fixed and available in the 5.0.10i version (http://snomone.com/d...ion-5.0.10i.xml).
After upgrading, remove the MAC and save them again.
So - I hope I understood what you meant. This is what I did.
I upgraded to ver 5.0.10i (CentOS64), removed the MAC from the extension and saved. Re added the MAC into the ext. Re provisioned via MAC addr. Its still only pulling up 1 line.
I did the above steps again, but this time I removed the snomone/generated/pbx-domain/extension file and I got the same result. 1 registered line.
Any more suggestions?
. So adding the stanza should result in the same .cfg as it is now. 
lol.
Haha - Can that be our 'Get out of jail free" card? Just tell the customer that big brother is messing up their encrypted sip traffic. 
Set Caller ID Text
in General Setup
Posted
Anyone?