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olecoot

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Posts posted by olecoot

  1. The problems with cross domain or inter-domain dialing seems to still (at least for me) be in version 3.4.0.3201 (Linux).

     

    My scenario:

     

    We host multiple domains on our servers.

    Each domain has it's own set of trunks, some with many trunks, some with few.

    None of the dial plans are global.

    Loopback detection is set to off.

     

    I have followed the suggestions in the forums and in the knowledge base to use "Try Loopback" in the dial plan. They don't seem to work in my case.

     

    Is there anyone who has a scenario close to what we have that is able to call from domain to domain on the same physical server? If you have, does someone care to enlighten me on how it's done?

     

    Thanks in advance

  2. Any progress on this topic? There are many of us who cannot get inter-domain calling to work for various reasons. There are also those of us that do not configure our servers with a single "common" trunk that all of the domains use. For instance, in our case we have multiple domains per server with each domain having it's own trunks.

     

    Well you guys are getting much further than I have been able to at this time.

     

    Running 3.3.1.3177 (Linux), alias set on hunt group 113 in B.domain.com to 5554443333

     

    Dial Plan in A.domain.com has one entry Try Loopback match * with null replacement.

     

    Debug going and the log tells be the following.

     

    [9] 20090706153623: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:\1@\r;user=phone!i against 5554443333@A.domain.com

    [8] 20090706153623: Dialplan TEST: 5554443333@A.domain.com not found in the local system

     

    I read this and say to my self - of course 5554443333@A.domain.com is not found in the local system, because it is 5554443333@B.domain.com.

     

    These types of changes is what keeps us running 3.1.0.3031 (Linux). If it's not broken, why fix it...?

     

    I think it is going to be easier to figure out how to get my sofswitch vendor to change their code than to make changes on every server and domain dial plan whenever or if ever we figure out this 'new' method for inter-domain dialing.

     

    Right now - leaving it setup the old way on this version the log file just states - [5] 20090706162813: Received loopback request without tag.

    If I could figure out what this highly detailed debug message means - I could probably get the softswitch vendors help.

     

    Any ideas on which tag this message talks about? Could if be the ftag in the Record-Route header? Does this need to have a different tag id? Or does the the tag of the from header need to change when coming back to the PBX?

     

    Thoughts?

  3. For that you need to have the recording license (part of the call center editions). Then the web interface will show the options to record calls.

     

    Our license does have the recording option. The following options are available at the domain level:

     

    Record incoming calls from hunt group

    Record incoming calls from agent group

    Record incoming calls from extension

    Record outgoing calls to internal numbers

    Record outgoing calls to external numbers

    Record outgoing calls to emergency numbers

     

    What this particular customer wants is to

     

    Record incoming calls to internal numbers (or DID). Can this be done?

     

    I have an option at the Account level that reads: "Record incoming calls to extension" but this option will not record and incoming call from an external number or an internal number.

     

    I am on version 3.3.1.3177 CentOS.

     

    Thanks in advance

  4. How do I setup recording of all incoming calls to internal numbers or DID without using a * code. We have a customer that insists on having this feature and not have to use a *12 to accomplish it. Is this something that can be done presently? Not all extension will be part of an agent group or hunt group.

     

    Thanks in advance.

  5. I have BLF working with Polycom phones and PBXnSIP. One issue that I have is the ability to monitor when a phone is on DND via the * code (*78). What happens is when you enter *78 on the phone the extension will light on another phone while the DND message is played back. As soon as the message is completed however, the LED on the monitoring phone goes out, because the call completed. At this point the phone being monitored shows no indication that it is still in DND.

     

    I know that there is a way to do this with Snom phones (http://forum.pbxnsip.com/index.php?showtop...p;hl=star+codes) but have not been able make this work on the Polycom phones. Has there been any advances in this area?

     

    I am using PBXnSIP version 3.1.0.3031 (RedHat Linux) and version 3.1.3.0439 on the Polycom phones.

  6. I think there still may be an issue.

     

    If I set the recording options for the domain to off, then go to an extension and enable recording for the extension (to override the defaults for the domain), no recordings happen. If I enable the recording options for the domain, everything works as planned.

  7. I may have something mis-configured but I'm having an issue recording calls on version 3.2.0.3143 (Linux). Recording defaults for the domain are set to no. On the account that requires recording, I have the recording settings "Record outgoing calls to external numbers on", "Record outgoing calls to internal numbers on". Destination for recordings is "recorded-calls/$m/$d/$u-$t-$n.wav". However, recording did not work with the default location either. This worked on a previous version without fail. I can't seem to get it to work on this version with the settings mentioned. Linux is RedHat EL5.

     

    Using *12 and attaching recording to email does work.

     

    Thanks in advance.

  8. The new codec selection field is just a glorified JavaScript input form for the gool old list. If you look at the pbx.xml file, you will still see the old list of codec numbers. The change was made to make it easier to select the codec preference.

     

    If there is no codec selected (e.g. on the trunk) then that means that the system default should be used. Only in the admin/ports web page there must be at least one codec selected (because that is the default codec preference).

     

     

    So why is the codec list on the admin/ports web page empty? And how do we populate it?

  9. I have a question on the codec settings field in the System Admin >> Ports >> RTP >> Codec Preferences. The list here is not populated with codecs and does not give you the ability to do so (see screen shot attachment). How does this field relate to the codec settings in the pbx.xml file, namely <codec_preference>0 8 9 18 2 3</codec_preference>?? Are the settings in the pbx.xml file used even though they do not show in the System Administration Web Interface? Are the codec settings that would be listed in the System Admin >> Ports >> RTP >> Codec Preferences fields downstream, upstream or both? Also how are these settings (pbx.xml) different from the Override Codec Preference settings in the Trunks?

     

    Thanks in advance.

    post-171-1235574065_thumb.jpg

  10. Aside from the financial aspect of the move, would you share your reasoning? We have deployments using both, and find that we have far more options managing our windows installations. Performance and reliability has been no issue on either platform.

     

    We are a hosted environment, meaning multiple domains on multiple servers. The main reason behind the migration is that we have found we can have more domains per server on the Linux servers with less impact than on the Win2003 servers. Please see the attached pdf.

    Media_CPU_performance.pdf

  11. clarity, Have you been able to accomplish this?

     

     

    Looks like the problem(s) are that the examples given on that site do not work.

    I'm guessing the web interface tries to do it the same way which does not work.

    Further looking around shows that you have to issue the command like this:

     

    The -c and -p must come FIRST then the arguments.

    *this works*

    pbx:/srv/pbxnsip# taskset -c -p 0 3330

    pid 3330's current affinity list: 2

    pid 3330's new affinity list: 0

    pbx:/srv/pbxnsip#

     

     

    Doing them the way that website suggests:

     

    # taskset -c 1 -p 13545

    Absolutely does not work in my Linux installation as well as some others I had read about.

     

    I'd still like to be able to set it from the web interface..

    Is that hard coded? or does it fire off an external script that I can 'fix'? ;-)

  12. Hi,

     

    We would like to migrate our PBXNSIP Windows installation to a Fedora Core 1 host.

     

    I asked our provider if it's working and this is what he said :

     

    Fedora Core 1 is compatible with Red Hat and CentOS, but it is recommended that you install a Fedora Core 1 package. Try on http://rpmfind.net/ and http://rpm.pbone.net/

     

    Does anyone have experience with this kind of migration?

     

    Can I backup/restore all settings, files, voicemails, autoattendants, etc... from the Windows version and restore in the Fedora version?

     

    What things should I consider before doing this?

     

    I am currently in the process of migrating our Windows servers over to Linux. After the Linux server is built, we have had no issues migrating the customers on the Windows servers over to the Linux server. We simply use the domain backup utility to do the migration. The only thing we find that does not get properly tarred up in the domain backup is customized MOH files that the customer may have been using but this is easily overcome.

  13. When creating accounts as an extension, can multiple email accounts be listed in the Email Address: Field, such as accountname@domainname.org; accountname2@domainname.org; accountname3@domainname.org; etc.

     

    The logs of my email server show error messages that state "<<accountname@domainname.org>... Unbalanced '<' (hold)

     

    I understand that the email program is adding the < but is the Unbalanced error due to the multiple email addresses in the field? If so, is there a work around (other than creating mailing lists)?

     

    Thanks in advance.

  14. how do I turn off tls on a domain by domain basis? Possible?

     

    Don't know if it is possible on a domain basis .... but can be done globally.

     

     

    In the pbx.xml file look for smtp_starttls. It defaults to true set it to <smtp_starttls>false</smtp_starttls> and restart pbxnsip. Worked for me.

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