similar to: Voicemail Issue - Failed to lock path

Displaying 20 results from an estimated 900 matches similar to: "Voicemail Issue - Failed to lock path"

2006 May 15
3
How to tell if RTP stream is has been reinvited?
Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com
2006 May 31
1
Can you dial with different CID's?
Is it possible to dial more than one extension with a different CID to each extension? I'm thinking macros might be needed, but I don't have a good handle on macros. Is it possible? Any hints? BTW - this would be used for showing an internal extension to one phone and a PSTN accessible number to another phone. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road
2006 Dec 18
1
Cisco 7940 - NAT Option
I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used.
2007 Jul 05
2
REGEX expression for NXXNXXXXXX?
Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com
2006 May 26
0
Sip Notify cisco-check-cfg - Does it still workwith 8.2?
It does on my test phone. Is your tftp server available? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brent Torrenga Sent: Monday, April 17, 2006 11:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still workwith 8.2? Has anyone else noticed that
2006 Jun 06
1
SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
Dear list (and more specifically Bret), I am getting one-way (inbound only) audio when trying to place a SIP call via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out "== Forcing Marker bit, because SSRC has changed" 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and
2006 Jun 14
3
Directory - First Name/Last Name - How to use both? a@h?
I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can
2006 Apr 10
0
RE: still no solution for me, if one
>Brent, > >you mean, I could just remove the remark signs and number it 103, 104, >105, .... since it does not matter why it failed (busy, congestions) >(maybe for statistic purpose to add a log entry for the move to the next >provider). > >bye > >Ronald Yup. Take a look at the macro solution, too. I don't fully understand macros (I'm no programmer), and
2006 Apr 17
0
Sip Notify cisco-check-cfg - Does it still work with 8.2?
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg doesn't elicit any response from the phone using fw 8.2? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com
2006 Apr 18
0
Re: Cisco 7940/7960 SIP 8.2 Freely
It doesn't seem as much broken as just annoying. I am holding off on upgrading until this resolves, but it doesn't seem to affect performance, anyways. BTW, some folks say that the server address only gets appended to the CID when a redirect or something comes about. Our experience here shows that the IP always gets appended. >Alexander Burke wrote: >> Just in case anyone here
2006 Apr 28
0
DNSMasq - Why the stuff hits the fan when the net connection is down
List, Per someone's suggestion (thanks, whoever you were) from this list, I implemented dnsmasq to prevent the issue of resolving DNS when the net connection goes down. This morning the net connection was down, and our * server didn't miss a beat. I recommend looking into: http://thekelleys.org.uk/dnsmasq/doc.html And http://www.enterprisenetworkingplanet.com/netos/article.php/3377351
2006 May 23
0
Sip.conf: domain=huh?
So I saw mention of a way to allow dialing using SIP URI's on Dave McNett's site at http://slacker.com/~nugget/projects/asterisk/page7 Wow, awesome, I can call anywhere now. However, I think there is a more elegant way of figuring out whether or not the local * server should handle a given domain. Specifically, Dave compares a series of domains within extensions.conf to figure out how to
2007 May 14
0
How is Context Determined when Transferring a Call?
When trasferring a call, how is the context determined? When using a zap device, and the DTMF code for blind or attended transfer is entered, does the tranfer originate at the context the zap device is set to be in, or does it originate from where the outside call being transferred originated in, or the context the current call is in? I ask because I am seeing strange behavior when trying to
2007 Aug 01
0
Can you specify a sip UA's codec based on IP?
Does anyone have any tricks to use some logic with SIP UA's codec negotiation based on the UA's IP? What I would like to do is have Cisco 7960's use g711u when they register with a local IP, and g729 when they register with a non-local IP. I was thinking about sip.conf and making two entries for each UA, one where the host=dynamic, disallow=all, then allow=g729; the other
2006 Apr 11
4
Why is the internet connection important to LAN and PSTN calls?
Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay
2006 Jan 30
1
Need to recompile * after changing zap echo method?
Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A.
2006 Feb 27
2
Echo on PRI/BRI?
Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com
2006 Jan 17
0
RE: Building from scratch would like the benefit of (TOO LONG...)
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very nicely. Also, the SCCP channel for * is under heavy development, and may offer a future option to convert in that direction, too (SCCP, or skinny, is their native tongue, not SIP). We got our phones from John Putnam at Global Technology Solutions - competitive price, very prompt service, and delivered them with the
2006 Feb 16
2
Random Hangups/Disconnects
Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have
2006 Mar 16
1
MeetMe - Causes * to crash :/
Anyone ever seen MeetMe cause * to crash? Specifically, it happens consistantly if someone begins to enter a conference and then decides to hangup while Allison is introducing them - like playing back "conf-onlyperson". This has been seen with the MeetMe participant connecting via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't seen it). The box is *