similar to: call from cisco router to asterisk gets auto attendant

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some

hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in cisco router, every thing will be ok. is there any difference between g711 codecs which cisco and asterisk utilize? On Wed, May 6, 2015

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

Hello; I have a SIP trunk with a service provider, the caller from landline or mobile is hearing us very well, but the agent who is sitting on the handset is not hearing well, the voice at the agent is not crystal (like he is talking from well or far deep place). Although the IP Phones are cisco 7942G and the used codec is g711ulaw (actually it gave better quality than g711alaw). If we increase

We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

What should my allow= line look like in h323.conf for G.729? I've tried allow=G729A but this doesn't seem to be right. These "codec indentifiers" sure are mysterious. Take g711alaw. To allow it you seem to have to use allow=ALAW. Even though "ALAW" does not show anywhere as an identifier when you say show codecs.

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback

Hi all, Firsts of all, let me say that I'm new to asterisk. I have some time suscribed to the list reading a lot of your messages and trying to learn a lot. The case is: last week I installed an asterisk server in the following scenario: PBX --- CISCO_ROUTER ---- ASTERISK The calls that are routed within the asterisk work perfect, there is not problem. However, the calls that are

I am testing asterisk-1.2.1-15 on RedHat 9(i386) for SIP-to-SIP call and i found a strange problem. When an extension gets a ring and it picks up the call a "tick" sound comes at start. This happens on both sides. I tried Xten's softphones and also hardphones. A thing which was common in both(soft/hard phones) was the selected codec. When i used g711ulaw on both soft/hard phones

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3). Well, the trunk is partially working, asterisk' extensions talk with cme, but - - when cme try to connect to asterisk' number, receives "the number dialed is not in service". - - calls from ISP through asterisk to cme don't work completely,

Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.