similar to: sip channel error - extension pattern matching problem

Hi, I am trying to utilize MWI with sip channel. when my client sens a SUBSCRIBE to asterisk I get info that user not found: <-------------> [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to

I'm a bit confused about how * constructs the RURI when it generates a BYE. For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs it from scratch. This

Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Hi, I'm trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP

Hi, I was trying to compiles addons 1.4 and res_config_mysql doesn't compile. is res_config_mysql still supported and is it still posible to use mysql with asterisk RealTime?? Bests Tomasz

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Hello, Is there any way to identify an incoming session based on the Diversion header? In my scenario, I have some unregistered endpoints (mobile phones) that make calls through our Asterisk, which controls the external call rights based on the endpoint's context. In a normal call, their number will be in the From header and the destination in the To an RURI, but when they make a call

Hi, I have the following situation. I have one account created in my VoIP provider. Asterisk registers this account with the usage of 'register = ' command in the sip.conf file. I have a number of aliases assigned to my user which correspond to different public/PSTN numbers through which I am accessible. When there is an incoming call from my sip provider 'some_extension' which

Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten => some_exten/calling-user.... is there some flag which activates this extra check?? Cheers Tomasz

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

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]

Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b

Hi, I'm using the Samba RPM's from Fedora Core 2 RPM's (3.0.7-2.FC2) as an authentication backend for a Squid Proxy server. It all seems to work fine, until I (try to)authenticate against a domain-group.. I started trying with 3.0.6-2.FC2, which also didn't work... This is a pretty clean/fresh installation of Fedora Core 2, for whatever that's worth... I've succeeded

I must be stupid because this cannot be this hard. I''m trying to build an all day calendar event. In most clients it shows up as an entry at the top of the day rather than blocking out the whole day with an event. I have a bit of code that looks like this... cal.event do dtstart DateTime.parse("#{startDate.year()}-#{startDate.month()}-#{startDate.day()}") dtend

Send CentOS-announce mailing list submissions to centos-announce at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos-announce or, via email, send a message with subject or body 'help' to centos-announce-request at centos.org You can reach the person managing the list at centos-announce-owner at centos.org When