similar to: Call group channel limits

Good Day All, I am experiencing some weirdness using the E&M channel and hope you can offer a little assistance with the problem I am having. 1) call comes into channel 25 (Second Span first channel of a Digium Quad PRI from SBC-PRI) 2) Call is sent to channel 1 (First Span first channel on the Digium Quad PRI connecting an ADTRAN via E&M Feature Group D) 3) Between rings one and two

hi list, how can i organize several pcs installed with asterisk and e1 cards to be seen from an asterisk server as one? so if there is a voip call that needs to be forwarded towards the pstn the asterisk server should find a pc that has free channel on it's e1 cards that is connected to the pstn side. /-- Pc1 - E1-\ VOIP -> Asterisk--- Pc2 - E1--PBX->PSTN

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

[This email is reposted without the log attaced. I have added more information to the end of this email as well] CVS head (unmodified) current as of today. We have asterisk connected to both the pstn and our pbx via two E1 pri connections. We use overlap dialing to cope with different number lengths in Sweden. Incoming calls work perfectly. Outgoing calls seem to not get the repsonse it

>> Hi! >> I have an Asterisk Box with one E1. This is connected >> with PSTN. My problem is that periodically the >> Asterisk console shows the following message. >> >> -- B-channel 0/1 succesfully restarted on span 1 >> -- B-channel 0/2 succesfully restarted on span 1 >> [..etc...] >> >> I don't know if this behavior is correct.

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Hi! I have an Asterisk Box with one E1. This is connected with PSTN. My problem is that periodically the Asterisk console shows the following message. -- B-channel 0/1 succesfully restarted on span 1 -- B-channel 0/2 succesfully restarted on span 1 -- B-channel 0/3 succesfully restarted on span 1 -- B-channel 0/4 succesfully restarted on span 1 -- B-channel 0/5 succesfully restarted on span 1 --

Hi Leonardo At first should be useful to post your message at asterisk-r2-request at lists.digium.com group. By the other way let me advice, to make an explained detail of your problem as; Asterisk version Openr2 version Configurations files Dialplan dahdi pattern detail Detail of the call process (inbound or outbound call failed?) In case of outbound call failed extension dial wait time of

Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciateany idea. --------------------------------- Moody friends. Drama queens. Your life?

Hello everyone. I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri 1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from telco. Another trunk looks to PBX with DECT system. Some outgoing calls from asterisk to PSTN drops. The last message that exists before hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This

Hi, We have integrated an Asterisk system (built from CVS HEAD) with an Avaya IP Office switch running Q.SIG on an E1 interface, using a Digium Wildcard TE110P. The system is being used for handling both inbound and outbound call traffic to a set of Asterisk agents who are connected via extensions on the IP Office - there is a maximum of 15 agents at any one time, allowing at least another

https://bugzilla.mindrot.org/show_bug.cgi?id=1424 Iain Morgan <imorgan at nas.nasa.gov> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |imorgan at nas.nasa.gov --- Comment #36 from Iain Morgan <imorgan at nas.nasa.gov> --- Created attachment

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Hi, Let us know more information about your setup. Hardware/software details details such as. server configuration PSTN cards you are using?? ( E1 or FXO card) sip.conf, zapata.cons, zaptel.conf config details?? Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased ================== Message: 5 Date: Mon, 10 Dec 2007

On 6/18/06, Iain Haslam <iainhaslam at gmail.com> wrote: > > BTW, I really like the new > > > > ```source > > code > > ``` > > > > syntax. *Huge* timesaver. > > Continuing this particular off-topicness: Lucas, are you aware or in > favour of the bc(code). syntax already being used for this [1]? I > mentioned it previously on the list [2],

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Hi, would be glad to be contributing with this question to all the comunity. I?m having a weird issue, suddenly I get channels on IDLE 0x00 state until I do a dahdi restart. Trying to do a call trace to see whats going on deeper, I get surprised when tried to open the .call file to see the log this is incomplete. This is what I see [root at localhost telefonica]# nano

Hello Every one I need to enable Asterisk to route external land line calls to the PSTN. Regarding our environment, we have Cisco CallManager (3.3.4) to which IP phones are connected. E1 terminated on a couple of As 5300's which are controlled by a soft switch (Cisco PGW200 Call Control). What is the best scenario to route external calls to PSTN. Should I use SIP or just connect Asterisk

Hi all. I have this problem with my Digium 2E1 card and PRI, for hours It works well, with some meesages... [Aug 10 09:20:31] NOTICE[32270] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 But PRI continue up....hours later... PRI go down. I thought the problem was in the telco, but the strange thing is that I have a loop cable in the second E1 and when I scan both E1