similar to: Traffic Crossover

Hello all, I have a project which requires me to rout calls from ten blocks of sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers

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Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

Is there a way to disable the b2bua feature in asterisk. I would like asterisk to work as a sip server and not be involved in the RTP path between phones. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/3716b8e2/attachment-0001.htm

Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31

Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten => 12345678,1,Answer() exten => 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -----Original Message----- From: Erick Perez [mailto:eaperezh at gmail.com] Sent: Thursday, July 26, 2007 7:03 AM To:

Hi again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

Hello, Have you tried the wanpipe-beta8c-2.3.3.tgz release in the custom/2.3.3 dir on their FTP site? Also, have you contacted Sangoma for support? They are very responsive. I am using wanpipe-beta8a-2.3.3.tgz and it's been working great on my A104 for a week now. MATT--- -----Original Message----- From: Dmitry Zhukovski [mailto:DZH@comx.dk] Sent: Monday, May 09, 2005 5:20 AM To: Asterisk

Hi, What are the current best practices for running asterisk as SIP B2BUA? Are there any sample configs online or the books that detail this configuration for the newbies? I'm going to run it behind 1:1 NAT for the clients in the public internet so I will use the externip, localnet, and nat settings. Thanks, Andrew

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

Hello, in our company we are trying to do this: Fax <--> Traditional PBX <--> Asterisk <--> PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax

Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

Hi everyone, any chance * could be used as a b2bua without forcing the media stream through the same box? I would love to do some computing on incoming calls, do things like setting another callerid and the forward the call to another sip UA - all without any audio traversing the * box. Any ideas? Thanks, Thilo

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions and critics welcome! http://b2bua.berlios.de Best regards, Mike

Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions welcome! http://b2bua.berlios.de Best regards, Mike

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

Ok, I have tested with almost all versions both in 2.3.2--*-stable and 2.3.3-*-beta - I am getting same messages: May 9 10:55:26 WARNING[3961]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! and same Down state pb01*CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7

Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Is the asterisk B2BUA patches useful anymore?? I'm trying to set a prepaid SIP network and the only way seems to get through a patched asterisk with B2BUA functions.. The patches failed, Hunk + problems: I have repaired them, but is it very useful?? Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG

I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK = NORMAL TE_REF_CLOCK = 0 wanpipe2.conf: TE_CLOCK = MASTER TE_REF_CLOCK = 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works.