similar to: Searchable Archives?

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54

Hi Folks, Can you point me in the direction of searchable archives for the list? I'm obviously missing something. I would like to be able to search the archives before asking my question. Thanks in advance Simon

Is there a searchable archive of this list? Did I overlook something obvious? I can find the archives, but short of downloading all the monthly gzips and building my own searchable database, it seems my only other option is to go month by month looking at subjects and hope to stumble on what I'm after. Does anyone maintain a public searchable version of the archives? I've got

Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is

Is there a searchable archive of the lists somewhere?

I was interested in going back to research past discussions on the list about possibilities for a syntax to add a class="foo" to div's and/or span's to Markdown text, but didn't have much luck. IIRC, the list never had a searchable web interface, but older posts were available on a Google search. This seems not to be the case any more. Can anyone recommend a good

Hi, Is there an easily browsable and searchable archive for cfe-dev? I often find myself wanting to link to a particular email, and it’s a very time consuming process with a pipermail interface. I’ve noticed that llvm-dev is reposted to a google group (https://groups.google.com/forum/#!forum/llvm-dev <https://groups.google.com/forum/#!forum/llvm-dev>) which can be super convenient for that

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

On 15 March 2018 at 02:21, George Karpenkov via cfe-dev <cfe-dev at lists.llvm.org> wrote: > Hi, > > Is there an easily browsable and searchable archive for cfe-dev? > I often find myself wanting to link to a particular email, and it’s a very > time consuming process with a pipermail interface. > I’ve noticed that llvm-dev is reposted to a google group >

I'm going to post a question to the group that I'm sure has been answered before, but all the googling in the world isn't helping me. Is there a searchable archive of this mailing list somewhere?

Hi there, our old pbx contract will expire soon and we have now 2 pbx companies trying hard to sell us a new VoIP one. I am usually a friend of open source software and read about the Asterisk poject, but I can not assess if Asterisk fulfills all our requirements. It would be great to receive some input from experienced users to be able to convince our management. My question is if the

Hello all, I am trying to connect asterisk to a Siemens Hipath 3750 PBX system. I have a physical connection issue. I know that I should use a crossover RJ48 cable to link the two systems. The problem however is that the physical interface of the Siemens system is very unfamiliar. From my digging around, I think that this is an S2M interface. http://www.mail-archive.com/asterisk-users at

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. Thanks Chris

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Hello all-- I can't seem to find a way to search the archives of this list right now though I DO remember doing that in the past. Any ideas? Thanks. -- Kay