similar to: G.729 with RVA

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:

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

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

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

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have "A" asterisk with numbering 45670 "B" asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |

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

I am now at a total loss. Using Sipura spa-2000s connected to *, I get DTMF working just fine for internal extensions, voicemail, etc. If making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial tone. I am working unsuccessfully with Cisco right now on this, but they cant find anything wrong. I have tried all suggestions I can find from the list and elsewhere.

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

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

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

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)

Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

Hi, Just wondering if anybody has encountered a similar problem as I have with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have dtmf relay configured on the AS, however, when someone calls in from the PSTN sometimes their digits are inputted twice, which messes up the extensions. If there is a better way to terminate calls from a AS without using SIP, that would fix this

Hi, i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco AS5350)and user is connected via sip too. When user calling out (via AS5350) he receives progress tone generated by voip-phone not that passing from telco line. I turned on debug and see that the AS send: 183 Session Progreess but to user is sent Ringing, not progress. I have progressinband=never in sip.conf so

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.

Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No

Hello All, I looked at the codec supplied by Digium and I think that it is G.729 Annex-B (G.729b - 8kbps). Correct me if I'm wrong. I have also looked at the article from Cisco "VOIP Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation", which states that G.729b is supported on pretty much everything except for AS5350/5400. Also that G.729b is a "high