similar to: Channels hanging when SIP phone gets reset during call

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

In version 1.2.7.1 I have an endpoint (number 5302) registered. 'sip show peer 5302' shows that the Reg. Contact address is: sip:line25@192.168.5.203:5066 When I call 5302 I see INVITE messages correctly routed to the contact address with request lines like: INVITE sip:line25@192.168.5.203:5066 SIP/2.0 But when NOTIFY messages are sent, the request lines are incorrect, like this:

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA.

Hey list, I'm having problems to get running *. I don't have any digium hardware yet. I just want to perfrom some tests using SIP. I compiled asterisk and zaptel with ztdummy enabled on Fedora Core 3. When I try to start ztdummy I get the following message: localhost# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line0: Unable to open master device /dev/zap/ctl 1

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

Hi, I am trying to benchmark disk I/O performance on VM running with "-snapshot" option enabled. In order to do that I specify cache=none in the -drive parameter (yes I am running qemu-kvm at command line0. The problem is that if I use this option kvm seems to ignore the cache=none directive and I get weird output from iozone (I/O values are better than on the real machine,

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two

Minor update - use the following: > if (strcasecmp(data, > "x-Asterisk-Request-URI-pseudo-header")==0) > { > ast_copy_string(buf, p->initreq.rlPart2, len); > -----Original Message----- > From: Steve Langstaff > Sent: 23 October 2006 09:58 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users]

Dear All, I wish to calculate the Gini index (ineq from same package) and some other indices for the diameter distribution of each plot (df dgtot). dgtot: IDPlot Diameter(cm) 1 4 34.0 2 4 23.0 3 4 38.0 ... 51 5 16.0 52 5 8.0 53 5 9.0 ... 5301 140 25.0 5302 140 12.0 5303 140 7.0 I use: >

Hello, I am looking for a "real life" example of using SLA lines under Asterisk. I'll describe my environment and would like to know how I define it in Asterisk (version 1.4 final). Suppose I have two multi lines phones. The first phone has extension 1 assigned to it, and the second phone has extension 2 assigned to it. Now, I want extension 3 to be available on both phones as

Dear Peter, My inbox is MDA_external Storage: 17GB of 24GB Subject / From / To is fast but FTS(Full Text Search) for body is horrible. I suppose this is where we need Apache Solr. Do you think my mail storage format is bad? Do I need to change for better performance? Please advise Kevin On Thu, Jan 29, 2015 at 12:25 PM, Peter Hodur <petehodur at gmail.com> wrote: > > * Kevin

I just recompiled my version from this morning's CVS Head. My systems voice files (voicemail, time etc) were playing nicely. Until that is I added an extension and now the files won't play. Worse than that, * thinks the files have played and goes to the next step in the dial plan. What gives? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.173 0022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.173 0022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.173

Hi, sorry for the double post, I stupidly composed this as a reply to an earlier mail, which causes it to appear in an older thread. Posting again so it doesn't get lost in the archives: dovecot 1.1.7 reliably crashes every time I try to open a mailbox using IMAP. Error in the logs: dovecot: Dec 02 23:14:15 Error: setmntent(/etc/mtab) failed: No such file or directory dovecot: Dec 02

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't