similar to: Asterisk/Realtime and MySQL

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

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 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 I have finished the installation of my VoIP basic configuration ... Actually: - All calls from my E1 are received by a Cisco AS5300 and sent to my Asterisk (in G711 by SIP). - All user are connected by SIP to the Asterisk - All calls from User are sent by asterisk to the Cisco AS5300 Now, i want see if i can supply T38 Fax Gateway .... I am search to: - Cisco Receive all

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 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

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Greetings, I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error. I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated.

All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the

Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do it? How? Do i need an special IOS version? Ive been trying to compile the OpenH323 channel for the last month, but errors still happens. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc

Hi everyone, I'm new to the list, so I apologize if this is a trite question. We're an ISP with a barely used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. We have a notion that we want to take on a new revenue stream, with our model being as such: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

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