similar to: Progress passing problem.

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Chris Abel writes: >Hello everyone! > >I've had this problem for a while and cant figure it out. When an outside >caller calls an extension on my asterisk system, they do not hear any >sort of ringing. Inside extensions calling other extensions do hear >ringing. We have 3 other asterisk systems that are configured the same >way, but do not have this problem. We think it

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

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Hi All, We have a Cisco CME linked to our Asterisk PBX (named 'epstein'). Off said PBX we have numerous other PBX's, some IAX and some SIP. On a call placed from CME (SIP) to 'epstein' it all works fine except for a few quirks. When calling through epstein to an IAX peer we get '100 trying' followed by '180 ringing' sent back down the SIP leg to CME.

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

I really hope that someone can help me with this one. DTMF tones are not working for certain places that I call, specifically 1-800-882-8880 which is the AA advantage line. It works for almost everyplace else. If I bypass asterisk, the call works fine. Network looks like: <SPA-2000> --SIP-- ASTERISK --SIP-- <AS5350> --PRI-- PSTN sip.conf entries [VGW01] (this is the AS5350)

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Just happy. hardware information: -------------- Some small factor IBM Celetron (coppermine) at 1100 (11*100FSB) 256 RAM 15GB Hard. 1 x Digium E100P - E1 Line from telco with 300 Dids 1 x TDM400P for local phones --------------- Few small machines (mainly brand PII at 233Mhz with TDM400P Cards. --------------- There is a lot of SIP equipment attached: 2 x Micronet SIP Gateways 1 x ata186 1 x

> Hmmm... I have this aweful feeling that I'm choosing the > exact wrong time to ask a "newbie question" :) Oh well, here > it goes. > > The quick question is : "How do I dial an extension?" > (answer is probably - "you don't" in which case:) "How do I > dial my asterisk box?" - I have no outside line, I just want >

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless "HDLC Bad FCS" errors. I modified logger.conf to get rid of the messages (so I could see what else was

Hello, I have question regarding the changes that are made in the sip protocol in Asterisk - the option progressinband. When this option is set to yes in asterisk version 1.4.21.1 - the call flow is: sip.conf: progressinband=yes Device Asterisk -----------INVITE SDP---------> <---------100 Trying------------ <-----183 Session Prgoress-- After version 1.4.2X+ (tested

Hi everyone, We have an Asterisk 1.4.17 user who has problems with sometimes not getting a ring tone on the calling phone. We're considering setting progressinband = yes, but would like to know how much extra CPU load this will require? If anyone can give something even roughly specific (eg "30% increase") that would be great, rather than just "lots". Also, are there any