search for: callrefer

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

Can anyone work out why my callerid doesn't work on my isdn4Linux with asterisk (or without asterisk for that matter)... This used to work fine, and I am quite confident that the telco is sending callerid information (because they always do on all ISDN lines standard, only extra cost on POTS lines). This is the information from dmesg, whether asterisk is running or not: isdn_net: Incoming

...Caller:834f9d0 h323pdu.cxx(494) H245 Sending PDU: request masterSlaveDetermination { terminalType = 50 statusDeterminationNumber = 4068038 } 1:21:35.143 H225 Caller:834f9d0 h323pdu.cxx(494) H225 Sending PDU: { q931pdu = { protocolDiscriminator = 8 callReference = 8176 from = originator messageType = Setup IE: Bearer-Capability = { 80 90 a5 ... } IE: Display = { 75 6e 6b 6e 6f 77 6e 00 unknown. } IE: User-User = { 20 a8...

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

...ed incoming call thread 5:37.445 H225 Answer:9cc1250 transports.cxx(1127) H225 Awaiting first PDU 5:37.470 H225 Answer:9cc1250 h323pdu.cxx(517) H225 Receiving PDU: setup 5:37.471 H225 Answer:9cc1250 transports.cxx(1136) H225 Incoming call, first PDU: callReference=27042 5:37.471 H225 Answer:9cc1250 h323caps.cxx(1942) H323 Added capability: UserInput/hookflash <1> 5:37.472 H225 Answer:9cc1250 h323caps.cxx(1942) H323 Added capability: UserInput/RFC2833 <2> 5:37.472 H225 Answer:9cc1250 h323caps.c...

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

...oming call thread 0:26.011 H225 Answer:40419ae0 transports.cxx(984) H225 Awaiting first PDU 0:26.029 H225 Answer:40419ae0 h323pdu.cxx(474) H225 Receiving PDU: setup 0:26.029 H225 Answer:40419ae0 transports.cxx(993) H225 Incoming call, first PDU: callReference=32 0:26.029 H225 Answer:40419ae0 h323caps.cxx(1626) H323 Added capability: SpeexNarrow-5.95k{sw} <1> 0:26.030 H225 Answer:40419ae0 h323caps.cxx(1626) H323 Added capability: SpeexNarrow-8k{sw} <2> 0:26.030 H225 Answer:40419ae0 h323cap...

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

...rver>==[10.0.0.0/24]==<altigen phone{8810}> It seems that the altigen is sending a ReleaseComplete and dropping the call before it gets routed to the snom190. Here's an h.323 trace 9: 0:26.489 H225 Answer:9a33350 transports.cxx(1136) H225 Incoming call, first PDU: callReference=1471 0:26.492 H225 Answer:9a33350 h323caps.cxx(1942) H323 Added capability: G.711-uLaw-64k <1> 0:26.496 H225 Answer:9a33350 h323caps.cxx(1942) H323 Added capability: UserInput/hookflash <2> 0:26.499 H225 Answer:9a33350 h323caps....

...c0 H245 Sending MasterSlaveDetermination 5:59.640 ThreadID=0x4cb6a1c0 H245 Sending PDU: request masterSlaveDetermination { terminalType = 60 statusDeterminationNumber = 4019430 } 5:59.641 ThreadID=0x4cb6a1c0 H225 Sending PDU: { q931pdu = { protocolDiscriminator = 8 callReference = 422 from = destination messageType = Connect IE: Bearer-Capability = { 80 90 a5 ... } IE: Display = { 32 31 33 2e 32 35 35 2e 31 39 38 2e 31 31 33 00 213.255.198.113. } IE: User-User = {...

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

Hi, I have about one year of experience with Asterisk, working with ZAP (digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite clear to me, the problem is that I have no experience with H323, but now, I need to use this also. The problem that I have is very trivial, so I think that this should be a very easy question for you guys whom know how it works. All I want to do,

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)