similar to: How can read the headers ISDN?

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

I'm completely new to asterisk and while we have access to Nortel experts none of them know asterisk and since I'm the network guy I've been lumped with this. This is what I'm trying to accomplish We have a CS1000 that's sip capable. I want to be able to connect an Asterisk box to the cs1000 via sip as an IVR, so I can pass a call off to asterisk, go through some IVR menus

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

Actually, there was recently a bug fixed regarding multipart SDP parsing in chan_sip. That should have fixed the issue with CS1000s and SIP (among other things). I haven't actually tried it yet on my CS1000, but it should work. Regards, - Brad ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Jerry Geis Sent: Tue 12/26/2006 3:51 PM To:

I am connecting to a Nortel CS 1000. I can place calls out to an extension so we are half way there. When calling into the box I get the following from sip debug ip X. I get dead air when calling into the box. In my sip.conf I have a context of nortel and in extensions.conf the nortel context just has a s,1,Playback(demo-congrats). Any suggestions as to why the call in might not be working but

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Hi, I have asterisk running a single T1 card with a connection to a nortel CS 1000. All calls to extensions, local and long distance are working just fine. My issue is this: The nortel CS 1000 supports connections to and intercom system that is just line level audio to speakers. When my PRI tries to call this "extension" it is never answered so my AGI never runs to play a message

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

Hi all Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial system , Thats the problem : Configuring ISDN BRI cards [A500/B700] ------------------------------------ No Sangoma ISDN BRI cards detected Press any key to continue: ------------------------------------ Configuring GSM cards [W400] ------------------------------------ No Sangoma GSM cards detected

Hi, folks! Sangoma has an informal user survey up on their home page (at http://www.sangoma.com) asking if people would use their A500 card for North American BRI, if it were supported. I encourage anyone with an interest in voice BRI in North America to vote; this information will be used for deciding whether to make the investment in developing a National ISDN driver layer for it. Benefits to

Hi, I've just been made aware of Asterisk through the article on Slashdot. We're a small UK company with a ISDN-30 link to the office and a Meridian (Nortel) PBX. We're expanding and moving offices soon and I'm looking seriously at VoIP technologies. So starter for ten, can Asterisk interface in with ISDN-30 technology? I'm assuming the set-up is a server sat running

I have a North American BRI configured as National ISDN (NI-1) on an SBC (Nortel?) switch. SBC calls this "FastTrak ISDN", and it's surprisingly inexpensive -- less than two POTS lines. I've been trying to find an inexpensive PCI interface solution to connect this to Asterisk as voice lines. I prefer ISDN BRI over POTS lines because of the improved signalling, fast dialing,

Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the

I would be grateful for some comments on our proposed machine specs for a new Asterisk installation at a client with an initial 70 extensions. The system should be able to handle 100 extensions. The system will have the following main features: - PSTN connection via ISDN 30, dealing with all incoming calls. Outgoing will be through ISDN initially - 70-100 Snom 300 handsets - 1-2

Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try to playback something I get the following error: **[WOOMERA]** HW DTMF supported s1c1- -- Executing [number at from-pstn:1] Answer("WOOMERA/g1/1-7b29", "") in new stack **[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29 -- Executing [number at from-pstn:2] Playback("WOOMERA/g1/1-7b29",

Hello, I'm looking for an ISDN BRI-U interface for use in the US. I'm primarily interested in using the BRI as a trunking interface into the PSTN with Asterisk. Naturally, cheaper is better. I currently use a Nortel Norstar system with BRI-U trunks, and really like the digital PSTN interface. Would really like to replace the whole mess with Asterisk but want to keep the ISDN trunks.

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Hello; we are looking to replace our current PBX with a *-box; it is connected to ONE ppp isdn connection that is terminated by the NC. We got on this box 4 msn's configured. currently we are working with pstn fxo's behind the PBX; it works but we can't use the CSID information behind it. We want to migrate and keep the MSN's to decide routing in combination with the CID.