similar to: Asterisk and CIsco 1760 SIP ?

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

Hi we have a Samba-Winbind 3.5.10 and i have a big quantity of errors in logs : [2012/09/24 16:01:29.248286, 1] winbindd/winbindd_ads.c:728(lookup_usergroups_memberof) lookup_usergroups_memberof ads_search member=CN=TESTDD,OU=Villeurbanne,OU=ExEA,DC=ELIOTT,DC=fr: Out of memory [2012/09/24 16:03:38.498533, 1] libads/ldap_utils.c:323(ads_ranged_search_internal) could not pull first

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi actually, i test a new Asterisk Server and i want add Mysql Realtime SIP. I read on the wiki: =================================================== Database Config put the following in res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = myuser dbpass = mypass dbport = 3306 Values in sip.conf or iax.conf like in older versions of * are no longer used. Database Table Lets

Hi i am not a expert on Asterisk and search a lot of small information : I use Asterisk 1.6.1.4 with MySQL. That's work and in my extension.conf, i have: [as5300-incoming] switch => Realtime and in extconfig.conf extensions => mysql,general,VOIP_Extensions A lot of Extension are into the table VOIP_Extensions. I am search to know if i can add a :

Hi I have a problems with a new Asterisk Server, when i want call, i have: [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160 handle_request_invite: Call from 'PHISIP000001' to extension '00420225352184' rejected because extension not found. but into my extensions.conf: exten => _00420X.,1,Set(CDR(CodeTier)=CZE) exten =>

Hi I have a problem with Asterisk 11.5.1. When I pick up an incoming phone call sometimes I need to transfer to someone else in the organization. I then dial a number on my phone, and press Xfer. Sometimes it works well, I mean, the number I dialed get the call and can chat with correspondent. Sometimes, the number I dialed get the communication and while he chats with correspondent, gets a bip

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

Hello, I am looking to know if it is possible to modify the SQL query that is on Realtime sip accounts. I want multiple servers use the same sql table, so getting an extra "server" field to indicate that the data is valid on the X server is this possible? thank you in advance jerome

Hi anyone know what is a the solution of this problems ? : [Nov 20 10:15:40] WARNING[12049]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to [Nov 20 13:04:45] WARNING[12043]: chan_iax2.c:9232 socket_process: I don't know how to authenticate Voip-Classic to 78.XX.XX.XX we have two Asterisk 1.6.1.4, this error are on the first server, used for the

Hi anyone know this error: Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0] sam/idmap_tdb.c:db_allocate_id(106) Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range full!! (max: 20000) Mar 20 12:01:06 gw winbindd[14756]: [2009/03/20 12:01:06, 0] sam/idmap_tdb.c:db_allocate_id(106) Mar 20 12:01:06 gw winbindd[14756]: idmap Fatal Error: UID range full!! (max: 20000)

Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following ------------------------------------------------------------------- exten => 802,1,SIPAddHeader(call_info=Classic-4) exten =>

Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes

Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.