similar to: sipconnect 1.0

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian

Hi out there, I think i've everything set up properly, outbound calls are working fine, but at incoming calls I can't hear anything, but the other one is able to hear me perfectly. I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to my sip-provider using a trunk. Firewall settings on the router are: forward UDP port 5060,5004,10000-20000 to asterisk server

I know I should probably post this to the Trixbox forums, but am hoping someone might have a quick answer here for me. Client with Trixbox 2.6.2 with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme conference with several people was in progress when the UPS died and the machine suddenly lost power. It is back up now, of course, but when you call into the conference room

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. > > I have a lot of endpoints and registrations on same SIP server. And it's > problem in pjsip now. Is not it? > > I

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote: > 07.03.2015 0:24, Kevin Harwell ?????: > > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Hello. >> >> Asterisk 13.2. >> I transfer configs from chan_sip to res_pjsip. >> In chan_sip i have

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using "BroadWorks". The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to asterisk 1.8.15.0. imagining in extensions.conf: exten => 1,1,Dial(SIP/121) exten => 2,1,Dial(SIP/121&SIP/122) When a caller dials extension 2 /and/ I have trustrpid=yes generaterpid=yes sendrpid=yes in sip.conf and I use the pickup exten, the caller is disconnected. see:

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

Hi all, Is anyone using oprofile? I'm getting segfaults from opreport at the moment, and I'm not sure if it is opreport, or just me. In case it is something just plain daft I am doing, here is how it goes: opcontrol --reset opcontrol --setup --no-vmlinux opcontrol --start ... now I run my program, /tmp/myprog ... opcontrol --dump opcontrol --shutdown then I run, opreport -l

Hello all If I look at the sip peers table definition as provided with the source of asterisk-1.8.23.0/ (looking at contrib/realtime/mysql/sippeers.sql) for the sendrpid column it's an enum with 2 possible values, yes and no. However, the sip.conf allows 4 values, no, yes, rpid and pai. Is this discrepancy an oversight? Is it possible to set the system default to pai but an individual peer

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =