similar to: SIP NOTIFY routing problem

Hi All. This is on Asterisk 1.2.13 I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes). I reset the phones (so they don't have time to say BYE). Asterisk seems to think that the call is still ongoing. This persists until I do a 'restart now'. asterisk1*CLI> show channels Channel Location State Application(Data) SIP/5301-089fc890

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,

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]

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

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.

Hello, how do achieve this: how must files /etc/sysconfig/network-scripts/ look like to be the same as entering the following two commands ... ip -f inet6 rule add fwmark 1 lookup 100 ip -f inet6 route add local ::/0 dev lo table 100 is there the localhost device lo correct, or does it have to be br0? e.g. a file route-br0 with 192.168.1.0/24 via 10.10.10.1 dev br0 does the routing to the

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

How do you get the called number on incoming SIP calls? I've never had multiple DID's via SIP from one provider before and somehow I never realized that with IAX it just works, and SIP is different. If I don't set an extension in the register command the incoming invite has <sip:s@me.com> in the To field. Now if I have multiple DID's that I want routed to different

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header

Hello there! I need some help to configure two Asterix boxes to route calls using SIP. I followed the instructions present at this site: "http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", but I couldn't get it working so far. The only difference, besides the names that I've used, is that I'm using realtime to retrieve

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.

Hi, I would like to set a sip phone for a user with fixed IP but I would like also to keep the user name in the invite, how is that possible? Is there any sip.conf setting that can be used for that? My sip.conf ... ; [rui] type=friend secret=oit host=127.98.12.88 canreinvite=no ; This situation generates INVITEs like this: INVITE sip:127.98.12.88 SIP/2.0 Via:

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

There's some one who can help me? I'm using Asterisknow with FreePBX and a Patton 4554 with 2 BRI ports on 2 ISDN lines. I would like routing the call entering by first BRI to one trunk and call from second BRI to another trunk. I have created 2 trunks both registering to Patton with different identities, actually all calls from both BRI are routed to one trunk. Thank in advance

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

hello ! My problem is: Astriks should create a connection to other members using a german Sip provider (www.sipgate.de). there are no problems with connections to: o Sip- Accounts o national phone numbers o mobile phone numbers but connections to international phone numbers DO NOT WORK (see the attached protokoll). The connection to international phone numbers does work when I directly use