similar to: Meetmee weirdness

I updated Asterisk to 1.6.2.7 and now the user introduction in the meetme application is no longer working: [May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel type registered for 'DAHDI' -- Created MeetMe conference 1023 for conference '800' [May 27 09:26:51] WARNING[2407]: app_meetme.c:3640 find_conf: No DAHDI channel available for conference, user

Hi all- My TDM long distance is provided by MCI. We use account codes where MCI sends a challenge tone after receiving 1NXXNXXXXXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this?

As the thread from the other mailing list he sent this to states, it is illegal to share the file(s) he is asking for. Below is the thread from the sccp users mailing list that he sent this to. <sccp mailing list> 2006/2/28, picciuX In fact: the one you mention is not a config file; it is part of the "Locale-Installer for Cisco Call Manager". You need a valid service contract

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hi- I have a 7960 using chan_sccp which gives me a busy signal as soon as I dial the 1 in my string of 915552223333. Can't figure out why. I do have a dialplan.xml file: <DIALTEMPLATE> <TEMPLATE MATCH="*" Timeout="5"/> <TEMPLATE MATCH="#..." Timeout="5"/> </DIALTEMPLATE> Anyone have any insights?

Hello, here is the SIP-log from my VOIP-phone when getting an MWI message from asterisk: NOTIFY sip:105@192.168.10.11:2054;line=g2kiz8tz SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33f02076 From: "asterisk" <sip:asterisk@192.168.0.1>;tag=as229cbc7c To: <sip:105@192.168.10.11:2054;line=g2kiz8tz> Contact: <sip:asterisk@192.168.10.1> Call-ID:

Commit 90805dc (c/s 26387:4056e5a3d815) ("EPT: Make ept data stucture or operations neutral") makes nested p2m tables be unlinked from the host p2m table before their destruction (in p2m_teardown_nestedp2m). However, by this time the host p2m table has already been torn down, leading to a possible race condition where another allocation between the two kinds of table being torn down can

I have an older Newbridge Mainstreet 3624 upon which I'm terminating some analog DID lines. They are effectively loop-start trunks with battery supplied by me (ie. FXO) and consumed by the serving central office. One major part of DID is the requirement for providing Answer Supervision in the form of battery polarity reversal on the analog trunks. Without it wierd things start happening, like

we have performance problems with printing with cups/samba we use samba 2.2.1a on this (print)server (sideproblem: the authentication to another samba 2.2.0a (file)server seems to be broken) first the printer's print postscript and we left it to cups (or ghostscript) to render the appropriate printerdriver. but this really torn down the maschine. now we have the drivers installed on the

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Once the system gets into this state, the only remedy is a forced power-off. What seems to be happening is that an NFS filesystem that auto-mounted over a WiFi connection cannot be unmounted because the WiFi connection is enabled only for my login and gets torn down when my UID is logged off. Any suggestions on how I can configure things to avoid this? I really don't want to expose my WPA2

Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway

On 08/30/2015 04:45 PM, John R Pierce wrote: > On 8/30/2015 2:20 PM, Robert Nichols wrote: >> Once the system gets into this state, the only remedy is a forced >> power-off. What seems to be happening is that an NFS filesystem that >> auto-mounted over a WiFi connection cannot be unmounted because the >> WiFi connection is enabled only for my login and gets torn down

Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes'

On Tuesday 06 October 2015 13:30:50 Richard W.M. Jones wrote: > This is just a straight refactoring. Various ad hoc string_* > functions that appeared in Common_utils have been renamed and placed > in the String.* namespace. The old vs "new" functions are: > > string_prefix -> String.is_prefix > string_suffix -> String.is_suffix > string_find ->

Hi Reid, It occurs to me that we should probably move the cppeh outlining tests, since they aren't actually X86-specific. We should probably also add an ARM+Windows run line to them. I'm torn between whether they should go in the test/CodeGen/CPP folder or in a new test/CodeGen/Win (or test/CodeGen/WinEH) folder. Thoughts? -Andy -------------- next part -------------- An HTML

to dovecot from UW imap. I think I will go all the way to maildir while I'm at it. I was having performance problems with mbox format files and converted to mbx but it's not really yielding a huge performance gain. Deletes and moves are still rather slow at times. my mailboxes are typically 2000 to 5000 messages. I've scanned the mailing lists a bit and picked up some hints on

Hi Ubuntu 12.04 LTS client with 3.6.3 joined to the Samba4 AD domain. smb.conf winbind enum users = Yes winbind enum groups = Yes idmap config *:backend=tdb idmap config *:range=10000-19999 idmap config ALTEA:backend=ad idmap config ALTEA:range=20000-40000000 getent passwd and wbinfo -u returns all AD users correctly wbinfo -g returns all AD groups correctly getent group fails. Only local groups

Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs